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A FOUNDATION 

FOR 

ELECTRONIC MUSIC 


2 ND EDITION 


Roland 


Contents Hi 

Contents 

Introduction v 

Chapter One: What is Sound? 

1-1 Introduction - • • 1 

1-2 Sound 1 

1^3 Pitch 1 

1-4 Resonance in Air Columns 2 

1 -5 Resonant in Strings 3 

1-6 Timbre 4 

1-7 Tone Color in Vibrating Air Columns 4 

1-8 Tone Color in Vibrating Strings 5 

1-9 Tone Color in Musical Instruments , , 7 

1-10 Loudness • 8 

1-11 Propagation of Sound Waves 8 

1-12 Echo and Reverberation 9 

1- 13 Spatial Effects 10 

T-14 Questions 11 

Chapter Two: Electronic Music 

2- 1 Introduction 13 

2-2 Live Electronic Music 13 

2-3 Musique ConcrdtB 13 

2-4 Recorded Electronic Music 14 

2-5 Waveforms 14 

2-6 Additive and Subtractive Synthesis 16 

2-7 The Voltage Controlled Synthesizer 19 

2-8 An Approach to Subtractive Syn^sis 19 

2-9 Noise , 20 

2-10 Computer Music 20 

2-1 1 Direct Synthesis 23 

2- 12 Questions 25 

Chapter Three: Pitch 

3- 1 Introduction , - ■ 27 

3-2 Pitch Relationships in Music 27 

3-3 Beat Frequencies 28 

3-4 The Natural Harmonic Series 29 

3-5 Consonance and Dissonance 30 

3-6 Pitch Standard . .31 

3-7 Exponential Progressions 31 

3-8 Cents 32 

3-9 Voltage-to-Frequency Relations - 32 

3-10 The Linear V€0 33 

3-1 1 The Exponential VCO 33 

3-12 The Low Frequency Oscillator ....... ^ 36 

3-13 Frequency Modulation 36 

3-14 The SasJe Symhesizer Patch -36 

3-15 Questions 38 


iv Contents 


Chapter Four: loudness 

4-1 Introduction 39 

4-2 Measuring Loudness ^ . 39 

4-3 Frequency Response of the Ear 40 

4-4 Dynamic Range ^-j 

4-5 Intensity of Harmonics 42 

4-6 Envelopes 44 

4-7 VGA Control Response 45 

4-8 Amplitude Modulation 47 

4-9 The Basic Synthesizer Patch , , 47 

4- TO Questtons 49 

Chapter Five: Timbre 

5- 1 Introduction 51 

5-2 Noise . . . . i 51 

5-3 Filters 51 

5-4 The Low Pass Filter 52 

5-5 The Voltage Controlled Low Pass Filter 55 

5-6 The High Pass Filter 55 

5-7 The Band Pass Filter and Band Reject Filter .57 

5-8 Effeotsof Filtering on Waveforms 58 

5-9 Resonance in Filters , 60 

5-10 Ringing in Filters 64 

5-1 1 The Ba^ Synthesizer Patch 64 

5-12 Questions gg 

Frequency Range Chart facing page 66 

Metric Conversion Table and Metric Relationships facing page 67 

Index 67 


Introduction 


As the title implies, the purpose of this book is to lay a 
foundation for the study of electronic music, particularly 
in relation to the voltage controlled synthesizer. Prxtical 
Synthesis for Electronic Music (published by Roland) should 
also prove useful to the reader. Through a series of practical 
exercises it shows how theory can be applied to a particular 
electronic music system. It also includes a step-by-step 
example of recording electronic music using a synthesizer 
and equipment found in most hi-fi systems. 

With the advent of smaller synthesizer systems and semi- 
professional studio equipment such as mixers and multi- 
^annel tape recorders, the world of electronic music is no 
long^ restricted only to large Institutions arnf organizations. 
For the creative musician, this opens up a whole new world. 
Not many unknown musicians get the chance to work 
with an orchestt^ to try their own arrangements or original 
impositions and the people at the record companies do 
not have time to work through a score to mentally "hear" 
what a new song sounds litce; they want to hear a demonstra- 
tion tape. Besides, there are many things which go into 8 
composition which simply cannot be written down on )Kq>er. 
Budding young composers seldom get a chance to hear their 
works performed by a full ort^estra or a large band^ and how 
are they to learn the art of Instrumentation If they never hwr 
the results of their efforts? By studyiiig ttia scoras of great 
music? By analyzing the sounds on great records? That is 
a little like studying great works of sculpture and then trying 
to make a statue while wearing a blindfold. 

The marriage of electronics and music opttis up a world 
limited only by the musician's imagination. The synthesizer 
can create all the sounds of conventional music and create 
sounds never heard Isefore. But without at least a basic 

understanding of acoustics and the synthesizer this world is 
unapproachable. This book tries to give that understanding. 

I would like to express grateful appreciation to my fellow 
workers at Roland without whose help and patience this 
work would have been impossible. 

Robin Donald Graham 
Synthesizer Project Manager 
Roland Corporation 

Osaka, Japan 
December, 1978 


What is Sound ? 


1-1 Introduction 

In dealing with methods for synthesizing sounds nnany texts, 
including this one, seem to emphasize the imitation of 
conventional musical instruments and natural sounds. 
There is an important reason for this. All synthesis is based 
on one simple principal: the building of sound from basic 
elements. To understand these elements, we must understand 
acoustics. If we learn to imitate sounds with which we are 
^nintliar we are well ion the road to being able to syn^esize 
any sound we can imagine, all the way from purely electronic 
waillngs, blips, and bumps to the creation of imaginary 
instruments, instruments which do not exist in the real woiid 
but which sound as if they actually did exist somewhere as 
acoustic instruments. 

The purpose of this chapter is to provide that essential back- 
ground \tt bmic acoustics. 

1-2 Sound 

Sound is what we experience when the ear reacts to a certain 
range of vibrations. These iribrations themselves can also be 
called sound. 

Most soLind sources produce sound by means of some part or 
parts which vibrate when rubbed, strucic, or excited by some 
other means. Others produce sound by direct excitation of 
the air as, for example, Ijlowinc] across the mouth of a 
bottle to set the air inside vibrating thus producing sound. 
Electronic instruments produce sound only indirectly; they 
produce variations in electrical current which are normally 
processed and passed through an amplifier where they 
are made stroig enbu{^ to dHve a speaker. The vibrating 
speaker cone then produces sound. 

The three qualities of sound are pitch, timbre (tone color), 
and loudness. It is the combination of these three qualities 
that allows us to distinguish between different sounds. 

1-3 Pitch 

Pitch is that quality of sound which makes some sounds seem 
"higher" or "tower" than others. Pitch is determined by the 
number of vibrations produced during a given time period. 
The vibration rate of a sound source is called its frequency; 
^e hiigher Hie ^equency (or the hl^er the vibration rate), 
the higher the pitch. 

Tuning forks are used to produce accurate test pitches for 
use in laboratories and for tuning musical instruments. A 
close inspection will reveal that when a tuning fork is struck 
on the surface of a piece of hard rubber, the prongs appear 
slightly blurred due to their vibration. Dipping the vibrating 
prongs Into a beaker of water will cause the water to be 
violently ^itated. 

In a similar way, the vibrating prongs of the tuning fork 
agitate the air around them, sending out sound waves. 


2 What is Sound? 


1-4 


This is shown in Fig. 1 -1 , When the prong moves to the side, 
it compresses the air particles together. This disturbance of 
air particles moves in such a way that a pulse of compression 
travels away from the prong. When the prong moves to the 
other side, it produces a rarefaction of air particles which 
move outward simltarly. 

It is important to realize that the Individual air particles are 

only moving back and forth, transmitting the waves of 
compression and rarefaction outward from the source. 

The distance between any two corresponding points in 
two consecutive pulses is called the wavelength (X). The 

wavelength will depend on how fast the vibrations of sound 
occur and the speed at which the waves travel. 

The speed of sound in air may be given as approximately 331 
meters per seccmd (or about 700 miles per hour). This speed 
will vary slightly with the temperature and density of the air. 
Neglecting these effects and assuming the speed of sound to 
be consent* the wavelertgtfl will be directly related to the 
rate of vibration of the source. 


v = V 


where: X = wavelength in meters 

V = velocity of vound In meters per second 
/ = frequency in vibrations per second 

Thus the higher tile frequency of the vibrations, the shorter 
the wavelength, the higher the pitch. 

Frequency is often measured in units called the Hertz (Hz), 
named kfter Heinrich Rudotf Hertz (1857-1894). If a sound 

source vibrates at a rate of 100 vibrations per second. It Is 
said to have a freciuency of 100 Hertz. 

The frequency response of the ear, or in other words the range 
of frequencies yiAt\dh can be heard^ will vary from person to 
person. The average person can hear sound from about 20 Hz 
to about 20,000 Hz. Some people can hear sounds as high as 
25,000 Hz. The upper frequency limit will drop, however, 
as a person gets older. For example, a person of 40 or 50 
years of age may be able to hear only up to about 15,000 Hz 
while In old age this may fall to 10,000 Hz. 

Resonance in Air Columns 

An analysis of pitch might begin with an Investigation of 
r^onance. It is welt known that the best way to keep a child's 
swing in motion is to give it small pushes in time with its 
natural period of swing. This is a classic example of resonance. 
The amount of energy exerted to keep the swing in motion 
is quite small compared to the amount of energy available 
from the motion of the swing. 

RettMtancB occurs whenever a body or system is set into 
vibration at its own natural frequency as a result of impulses 
received from some other body or system vibrating at the 
same fr^uency. 


Fig. 1-1 Prodiigtioh of Sound Waves 

Wave motion 


Particle motion 


n n 



— Pressure 


X - Wavelength 


Fie> 1^ Examples of PIteh Ranga* 
(See else chart facing p. 66) 


20 Average hearing 

20.000 

1^.4 Large pipe organ 

1 

8.372 

27.5 Grand piano 

4,186 

30.868 Orchestral music 

3,951.1 

1 I 
The extremely low organ pitch is exptained in Chapter 5 tp. 57]. 


What is Sound? 3 


1-5 


Resonance with sound waves can be demonstrated by using 
resonance tubes as shown in Fig. 1-3. By raising and lowering 
a tube placed inside a tall container of water, the effective 
length of the aif column Inside the tube is changed. Starting 
with a very short air column, a vibrating tuning fork is held 
over the mouth of the tube. As the tube is slowly raised, 
resonance will o&iur when the air column reaches a certain 
critical length and a relatively strong tone will be produced. 
If the tube is raised mon, other resonance points will be 
found. 

Fig. 1-4 shows whythis happens. Assume that the lower prong 
is moving down, as shown in (a). This will cause a pulse of 
compression to travel down the tube which will bounce off 
the surface of the water and return to the mouth. If the com- 
pression reaches the mouth of the tube just as the prong 
has reached its lowest position and is about to move upwards 
(as in {fo)), the compression will be followed by a rarefaction 
pulse which will also travel down the tube and reflect from 
the surface of the water. Up to this point, the prong has 
fihis*ied one complete cycle of wavelength of sound and 
the sound wave {one compression and one rarefaction) has 
travelled twice down and up the tube. It follows, then, that 
the leng^ of the column Is one quarter the wavelength of the 
sound. "I This is the principal which determines the pitch 
produced by all sound sources which consist of a tube with 
a closed end. 

Resonance may also be obtained with a pipe which is open at 

both ends, as shown on Fig. 1-5. In this case, the compres- 
sions and rarefactions travel the full length of the tube 
because there is no closed end to reflect the waves. The 
length of the tube will be one half the wavelength of the 
sound wave; or the wavelength of the sound wave will be 
twice the length of the tube.^ 

Resonance in Strings 

Three factors determine the pitch that a vibrating string will 
produce: Its length, its mass, and ihe amount of tension 

which holds the string stretched. As an example of these, the 
guitarist changes the length of the strings by pressing them 
against tiie frets to raise tfieir pitches; each guitar string is of 
a different diameter, the lower pitched strings being thicker; 
the guitar is tuned by adjusting the tension on the strings, 
the tighter the string, the higher the pitch. 

Experiments with vibrating strings may be conducted with an 
easily constructed monochord (sometimes called a sononw- 
ter), as shown in Fig. 1-6. 


Fig. 1-3 Closed End RmonanceTube 



■Glass or metal tubes 


-Water 


Fig. 1-4 Resonance in a Closed Tube 

® ® 



Compression 
down and up 


Rarefaction 
down and up 


Fig. 1-6 Open End Resonance Tube 


Y 


Glass or metal 
tubes 


1 . In actual practice, if measurements are made, a discrepancy in the 
tube length will be noted. This is due to the fact that the sound 
waves produced at resonance actually extend slightty beyond the 
open end of the tube as a result of air disturbance in this area. The 
amount by which the wave extends is called end correction. 

2, This statement neglects the slight deviation due to end correction. 
(Since both ends of the tube are open, end correction occurs at 

both ends). 


4 What is Sound? 


Resonance in a vibrating string can be demonstrated with a 
monoGhord and a tuning fork. The simplest approach would 
be to tune the monochorcT to unison wi^ the tuning fork 
by adjusting the tension on the string and the position of the 
bridges. If the unison is perfect, placing the stem of the 
Vibrating tuning fork on the point of one of the inn^^m hear 
the string will cause the string to vibrate. It is posslbis to 
find other resonance points of the string by moving the 
opposite bridge closer (witiiout hanging the tension on the 
string) and noting ^e positions where string vibrates. 

The frequency of the vibrations of a Stretched String is 
inversely proportional to iu length. Assumingthe tension and 
diams^ of the stringtemalhunchanged, if a string Of Hength 
(distance between the bridges) produces a frequency of x 
Hertz, then a string of I will produce a frequency of 2x 
Hertz, a string of / will produce a frequency of 3x Hertz, 
etc. 

The thicker and heavier a string is, the lower its frequency 
for a given length and tension. Thicker strings are usually 
stiffer which means that the sound will die away rather 
quickly, a feature not particularly desirable in musical instru- 
ments. To achieve the necessary mass without loss of flexi- 
bility, lower pitched strings are often wound with wire. This 
can be seen by examining the strings of a piano or guitar. 

It would be possible to present a similar discussion on the 
relation of size, tension, and mass to the pitches produced 
by other vibrating objects such as reeds, metal bars, etc. 


Fig. 1-6 Monochord 


Ideally, a pulley should be located here. For simple 
experiments where accuracy is not too imporunt, the pulley 
may be omitted, as shown. 



1-6 Timbre 

Timbre comes from French and means "tone color". In 
English, we pronounce it: tarn' bar or: tim'bar.^ 

Timbre or tone color is that quality of sound which allows us 

to distinguish between two sound sources producing sustained 

sound at the same pitch. 

Sound waves are the result of vibration. The vibrations of 
most vibratory systems tend to be quite complex so that 

they vibrate at several frequencies simultaneously. It is 
the combination and interaction of these frequencies, called 
overtones, which give a sound the quality we know as tone 
color. 


1-7 Tone color in Vibrating Air Columns 

In 1-4 above, it was shown how acoustic resonant points 

can be found using open or closed tubes and a tuning fork. 
Starting with the air column at its shortest, if the tube is 
slowly extencted inwe will be several points at vi^UA\ reso- 
nance occurs. The first resonance point will be stronger than 
the others with the next resonance points being progressively 
lower in intensity. Using ^e child's svving as an motogy of 
resonance, the secondary resonance points in the tubes 
would be like giving the swing a push every other swing, or 


3. a is in ^t;/ as in is; a ase in 490 


every third, fourth, or fifth swing, and so on. From this. It 
follows that if the impulses are not delivered every swing 
(assuming pulses of equal energy), the swings will be of 
lower intensity. 

Since various lengths of air column will vibrate in resonance 
to the one frequency of a tuning folic, it follows tiiat one air 
column of fixed length will form resonances at several fra- 
quendes. 

Assuming that the length of an open end pipe is such that It 
forms a resonance with the frequency of a tuning forte, the 
vibrating prongs of the tuning fork will act as Impulses to set 
the air inside vibrating. Since the tube is resonant at various 
frequencies in addition to that of the foric, the Impulses 
supplied by the fork will tend to cause the air inside to 
vibrate at all of these resonant frequencies simultaneously. 
These types of overtones are called harmonics because of 
their muiriGaHy harmonic relation to eadi othw. 

The lowest of these frequencies, or the fint harmonic, has a 

vtfavelength equal to twice the length of the open end pipe 4 
This frequency is usually the strongest and it is this frequency 
whKbh gives the total sound its overall musical piteh. For this 
reason, the first hannonie is also called the liindiumiital* 

The remaining harmonics are all multiples of the first har^ 
monic or fundamental. The next higher harmonic is tfie 
second harmoiilG with a frequency twice that of the funda- 
mental. Following that is the third harmonic with a frequency 
three times that of the fundamental and so on, theoretically 
Upwards to infinity. 

This series of harmonics is called the nitiml hannoiile 

series. Most vibratory systems tend to produce harmonics, 
but not necessarily all of the harmonics in this series. The 
sound from a closed end pipe, for example, conteins only the 
odd numbered harmonics. This is because for even numbered 
harmonics, the portion of the wave being reflected from the 
closed end is cancelled by fhB opposite portion of the wave 
coming from the open end. The tone quality produced by a 
closed end pipe, then. Is different from that of the open end 
pipe; it contains fewer harmonics and thus Is a little duller. 

Tone Color in Vibrating Strings 

In 1-5 above, it was shown how string resonance can be 
demonstrated with a monochord and a tuning fork. As with 
a fixed length of air column, a string with fixed characteristics 
will also be resonant at various frequencies, all of which fall 
Wi^in tite natural harmonic series. In the above mentioned 
string resonance experiment, if it were possible to obtain 
several tuning forks, each tuned to one of the harmonics 
of the first tuning fork, it would be seen that these fork^ 
would also cause resonant vibration of the string. As with ^ie 
open end pipe, ^e vibrating string can produce all of tfie 
harmonics in the natural harmonic series. 


4. Neglecting and correction, as before. 


6 What IS Sound? 


Rg. 1-7 shows three of the vibratory modes of a string 
(or three of the harmonics) as they would look if it were 
possible to completely separate tiiem from each other. In 
the fiindarnehtal mode of vlbratiion (a), nodes, or points of 
minimum agitation, naturally appear at the ends of the string 
since it is anchored at these points. In its next mode of 
vibration (b), aribther node appears at the center bf the stnng. 
Since this effectively cuts the length of the string in half, the 
frequency produced by this mode of vibration is twice that 
of the fundamental, br in other words, the second harmonic. 
The next mode of vibration (c) adds another node which ef- 
fectively divides the string into thirds, thus producing the 
l^ird harmonic. 

The tone cbtor etctuatly produced by a vibrating string will 

depend on how the string is excited and at what point along 
its length it is excited. Logic would dictate that a node could 
not appear at the point bf exdtation. If the string is ptucked 
or bowed at its center, it is impossible for any nodes (points 
of rest) to appear at this point, thus, since all even numbered 
harmonic$ produce nodes at the center, ihase harmonics 
would be missing from the sound. 

Antinodes,- or points of maximum agitation, for the second 
harmonic occur at a point one quarter of the way from the 
end of the string, as shovvn in Fig. T-7 (b). If the string is 

excited at either bf these points, the second harmonic wIH 
dominate the sound and any harmonic with a node ^ these 
points Will be missirtg from the sound (or in other words, 

the fourth harmonic and any multiple of the foLirth harmonic). 
Since the fundamental has neither a node nor an antinode at 
these points, it will be present, but weaker than if the string 
were excited at its center. It is possible to supressthe funda- 
mental entirely by lightly touching the second harmonic 
nodal point (the center of the string) wtlh edge of a 
card or feather, as shown in Fig. 1-8 (a). If the fundamental 
is completely suppressed, the musical pitch of the string will 
^em to be that of the second harmonic, since this Is nowthe 
lowest harmonic present. Other harmonics may be obtained 
with this same method using the related nodes and antinodes. 
f\g. 1-8 (b) shows how to obtain the sixth harmonic. The 
pitch produced will be two octaves and a fifth above the 
missing fundamental. IVIost string instruments are excited 
near one of the string, thus the sounds produced are 
rich in harmonies. 

A very similar discussion can be presented concerning the 
position of the pickup in relation to the strings of an electric 
guitar. The pickup detects ihe motion of the string pacing 
back and forth above it and generates minute electrical 
currents which electrically mirror the string vibrations. 
Assuming all harmonics to be present in the vibrating string, 
if the pickup is placed under the center of the string, it will 
be in the position of the node for the second harmonic. 
Since a node is a point of minimum vibration, the pickup will 
detect very little, if any, of the second harmonic portion of 
the string vibration. All haimonlcs with a node at this point 
will be missing from the output of the pickup. Other pickup 


Fig. 1-7 Modes of Vibration 
0 Fundamental mode 

A 

i 



N N N 


© Third harmonic 


AAA 

i I i 



N N N N 

N = Node (Point of minimum agitation) 
A - Antinode (Point of maximum agitnlon) 


FiB. 1-S Producing Harmonics 


® Second harmonic 



® Sixth hamionic 



positions will produce sound with different tone coloring. 
This is the reason that many electric guitars have more than 
one pickup. 

The material with which a string Is made, and its diameter 

will affect the tone color produced. Most string instrument 
players are aware of this fact; they often have favqrite brand 
name strings they prefer for use with their fnstraments. 
Thicker strings will be stiffer and therefore will have difficulty 
vibrating freely at higher frequencies. With thicker strings, 
the higher harmonics tend to die away very quickly. 

Tone Color in Musical Instruments 

Up to this point we have been primarily concerned with 

sources of vibration vtfhich produce sound, particulariy 

vibrating air columns and vibrating strings. The body of a 
given instrument also has a profound effect on the total 
tone color produced by that instrument. 

The solid parts of the body of aft instrument have their own 
resonances which are very similar to those in the vibrating 
string In that the frequency of these resonances depend on 
the dimensions, elasticity, and tensions of the various parts 
of the body. Many instruments such as violins, acoustic 
guitars, and drums also contain air cavities which resonate 
in ways similar to vibrating aJr columns. These body 
and air resonances all interact with each other and the source 
of vibrations so as to emphasize or de-emphasize different 
overtones produced by the vibrating sound source, and In 
some cases to completely cancel some overtones ar>d/or 
add new overtones not present in the original source of 
vibrations. This explains why two violins which look exaetiy 
the same may still sound quite different. For example, 
the wood used in the bodies may be different. 

As an example of the factors which decide the tone color of 
a musical instrument, let us consider briefly the factors 
which in general affect all wind instruments. First is whether 
1*»e air column is closed or open. Closed pipes cannot produce 
even numbered harmonics. Next is the scale of the pipe. 
This refers to the ratio of the length of the pipe to the 
diameter of the pipe. The larger the scale, the more difficult 
it is to produce the upper harmonics. Third is the shape 
of the pipe. Fourth is the intensity of the wind pressure 
used to produce the vibrations. Fifth is the method of 
excitation; in other words, whether by air passing across an 
edge, or by reed. Sixth is the nature and thickness of the 
walls confining tha iBr celumn. And last Is the size and shape 
of the mouth and related parts. 

Bells and cymbals are good examples of instruments which 
produce overtones that do not fall in the natural harmonic 
series. In the case of the cymbals, these non-harmonic 
overtones are so strong and numerous that they drown out 
any existing fundamental so that it is impossible for the ear 
to detect a specific musical pitch. Bells often contain enough 
harmonic overtones that our ears can recognize a musical 
pitch. The non-harmonic overtones are what give the sound 
its metallic "clanging" tone quality and is typical of sound 


8 What is Sound? 


sources which use metal plates or bars as a source of vibra- 
tion. 

One other point in connection with tone color should be 
meniSoned. The room Tn which a sound occurs will also 

affect the tone color of that sound. The room forms an air 
cavity with its own resonances which react to the sound. 
Smaller rooms, such as the average IMng room, often have a 
very strong effect on the sound produced in them. Especially 
affected are the lower frequencies. 

1-10 Loudnesi 

The first thing that may come to mind concerning loudness 
is perhaps dynamics (changes in loudness) in music. At first 
glance, loudness as a qudlW of sound might seem rather 
simple and not as important as tiie other two qualities, but 

this is not so. 

The loudness of a sound will vary during its production. 
This loudness contour is edited the envelope of the sound and 
often forms a very important clue as to the identity of the 
sound. Fig. 1-9 shows two envelopes. In (a) is shown the 
envelope for a single guitar note. As soon as the finger releases 
the string, the sound jumps very quickly up to its maximum 
loudness after which it begins to die away slowly. The actual 
maximum sound level reached and the length of time 
required for the sound to die away will depend on how much 
force is used in picking the string. 

In fig. 1-9 (b) is shown the envelope for a vocalized "ah" 
sound. With the guitar sound the basic shape of the envelope 
will remain unchanged; the only possible variation from this 
basic shape would be to dampen the string before the string 
vibration ceases. In the case of the voice, the envelope can 
be altered considerably, depending on the intonation of the 
singer or speaker. 

The relative intensity or loudness of the overtones contained 
in a sound also affect its tone color. The high overtones in a 
cymbal crash, for example, are very strong, much stronger 
than the lower overtones. 

1-11 Propagation of Sound Waves 

Sound spreads from the source in the form of a series of 
expanding spheres. As ^e sound waves travel away from ^e 
source, they become weaker in intensity. If we imagine a 
square measuring one centimeter on a side drawn on the 
surface of a balloon, we can see that the square will become 
progresively larger as the balloon expands. As sound waves 
travel away from the source, they must progessively cover a 
larger area, thus the sound energy is gradually dissipated until 
it finally becomes too small to detect 

Fig. 1-10 shows what happens to sound waves when they 
encounter an obstruction such as a wall. A part of the sound 
Is reflected from the sur^fae. Note the relation between 
the source of the sounds and the apparent source Of ■flie 
reflected waves. A part of the sound striking the wall is 
absorbed by the wall, and any sound whi(^ is not absorbed 


Fig. 1-9 Envelopes 

(a) Guitar envelope 

Loudness ^^^'^^'^'*^''^*»i„^ ^^^^^^^^ ^ 

0 .05 .1 .5 1.0 1.S 2.0 
Time (Seconds) 

(§) Vowel sound "AH" 


Loudness 


0 .05 0.1 0.5 1.0 1.5 2.0 
Time (Seconds) 


Fig. 1-10 Propagation of Sound Waves 

APPARENT SOURCE OF REFLECTED WAVES 



SOUND SOURCE 



What is Sound? 9 


or reflected will be transmitted by the wall to the air behind 

it. 

The amount of sound reflected, absorbed, and transmitted 
by the wait wfll depend on the materials used in its construc- 
tion. Hard smooth walls tend to reflect a great deal of sound. 
Shon/ier rooms and indoor swimming pools are good examples 
of rooms with highly reffectiwe walls. Soft porus materials 
tend to absorb sound. Heavy drapes and thick carpets are 
good examples of sound absorbers. Thin walls tend to 
transmit more sound than thicker walls. Most professional 
sound studios use heavy thici< walls to prevent transmission 
of sound into or out of the studio. 

1-12 Echo and Reverberation 

Echoes are produced when sound waves are reflected from a 
hard smooth surface such as a wall or cliff. If a person 
stands at a distance from such a hard surface and claps his/ 
her hands, the time elapsed before the reflected sound is 
heard will depend on the distance to the surface. In order for 
^e sound to be heard as a separate echo, it must be separated 
from the original sound by at least 0.1 second {100 milli- 
seconds). Assuming the speed of sound to be 331 meters per 
second^ the sound must travel a total of about 33 meters, or 
about 16.5 meters one way. If the surface is less than this 
distance, the returning echo will not be distinct but will seem 
to be a continuation of the original sound. 

A sound emitted within the confines of a room will expand 

from the source and bounce back and forth between the 
walls, floor, and ceiling of the room. Remembering that 
sound expands as a series of spheres, it can be seen that these 
multiple reflections will soon become very complex. The 
effect is that the sound will be reinforced and will seem to 
continue for a time beyond the point when the ori^nal sound 
source ceases emitting sound. This type of echo is called 
reverberation. 

Reverberation time is the amount of time required for the 
sound reflections within a room to die down to a given level 
after the source ceases emitting sound. The reverbera- 
tion time of a room will depend on the reflective qualities of 
the surfaces within the room and the placement of objects 
in the room. Some experts are said to be able to judge the 
size of a room merely by listening lo its reverberation charac- 
teristics. Many performers have been disconcerted by the 
seeming loss of carrying power they produce during a show 
as compared to reheasals in the same hall when empty. The 
audience and their clothing are good absorbers of sound. 

Many people Wke to sing while taking a bath. Most bathrooms 
have very good reflective qualities with long reverberation 
times. The reverberations tend to reinforce the sound of the 
singing so that even a person with a weak voit» sounds 
powerful and dramatic in the environment of the toth. 

In electronic music, it is important to differentiate between 
echo and reverberation. Echo is the effect produced by 
distinctly separate reflections of sounds. Reverberation is ^e 


10 What is Sound? 


effect produced by multiple reflections. In echo, the reflec- 
tions can be easily heard as separate repetitions of the original 
sound evefi when Ijiey overlap. This is the familiar echo 
effect obtained by shouting in the mountains. In reverbera- 
tion, the reflections are blurred together so tiiat distinct 
repetitions are not heard, but ratiier the orfgiriat sound sMitis 
to continue longer than normal. 

1-13 Spatial Effects 

In electronic musib, most sounds start out as electrfcat vibra- 
tions generated and controlled by the synthesizer and related 
studio equipment; therefore, it is usually quite "dead" 
sounding when put through a speaker. An artlffdti wiviron- 
ment must be added to the sound to give it life. This is one 
of the concepts of spatial effects. 

The most importanteffect in electronic music is reverberation. 
Wl^ reverberation we can create the feeling of intimacy felt 

with ;i chamber orchestra in a small room or the feeling of 
overwhelming power produced by the pipe organ in a large 
cathedral. The amount of reverberation controls the size of 
the space in which our electronic performance seems to take 
place. Not only that, but we are not limited to a one size 
"room''. We can change the apf>arent size of the space at 
will so that it matches the mood of the music at each moment. 

Reverberation can also be used to control depth or distance. 
The person who sits in the front row at the concert hall hears 
mostly the sound received directly from the instruments, 
while the person in the last row will hear more of the rever- 
berations than the direct sound (assuming speakers are not 
used). This gives a due to the control of dfstan» in electronic 
music. A melody played very softly with a great deal of 
reverberation added will sound very distant from the listener. 
A melody played loudly with little reverberation will sound 
very close to the listener. 

Besides special effects, echo can add complexity to music by 
giving slightly delayed repetitions of sounds. When used with 
sus^lnlng sounds such as string sounds. It can also add a 

great feeling of depth. The late repetition of these delayed 
sounds gives the effect of a much larger group of instru- 
ments. 

When a group of musfetans (or singers) ptay in unison to- 
gether, they will each be playing a slightly different 
pitch. This is one of the clues that tells us that we are listening 
to a group rather than a solo. Units designed to getmate 
chorus effects process the sound by altering the pitch slightly, 
then re-combining this with the original sound to give the 
effect of a group of players. 

P\\am i^lfHng is another common effect used In dectronic 

music, but the actual effect on the sound processed by a 
phase shifter is very difficult to describe with words. One 
deseriiption >nih\t^ o^es Mriy eUtxmt however, is to compare 
phase shifted synthesi^r ni^ia outputto the sound produced 
by a jet plane which hl^ just talcen off and is moving into the 
distance. Phase shifting is an effect which is also common 
with acoustic instruments and electric guitars. 


What is Sound? 11 


1-14 Questions 

1 . What are the three qualities of sound? Define them. 

2. Aiming the speed of sound to be 330 meters per second, what is the wavelength of a sdund wave with a 
frequency of 440 vibrations per second? 

3. What is the frequency response of the ear in the normal average human being? 

4. Define resonance. Give examples of resonance which occur in daily life. 

5. List three factors which affect the pitch produced by a vibrating string and show how changes in each will 
affect the pitch. 

IB. Mow are otferton^ produced and how do they affect the sound? 

7. Define envelope. Give some examples of familiar sounds and make diagrams showing the approximate 

envelope of each. 

8. What three things happen when a sound wave encounters an obstruction such as a wall. Make a sketch showing 
this. 

9. What Is the difference between echo and reverberation? 

10. Exi^aln how reverberation can be used to control the apparent distance between the sound and the listener. 


Words to define: 

antinode 

edhio 

env^lQpe 

frequency 

fundamental 

harmonics 

Hertz 

Hz 

loudness 
monochord 

natural harmonic series 
node 

non-harmonic overtones 

overtones 

pitch 

rarefaction 
r^sonan^ 
reverberation 
sound 

soundwaves 
spatial effects 
timbre 
tDn$ color 
wavelength 


Electronic Music 


Chapter TWo 


2-1 Introduction 

Throughout the ages composers have strived to break the 
bonds imposed by Iheir art. A study of musical scores will 
show how again and again composers have tried to break 
away from the conventions of their times in search of newer 
avenues Of expression. The use of newer hamiionfe structure 
and the invention of new instruments all came about as a 
result of this search. It is not surprising, then, that the 
discovery of electricity should usher in the developement of 
non-acoustic instalments. 

This text deals primarily with electronically generated 
(synthesized) sound. In electronic music we are not so much 
concerned w1l^ sound sources, but more with sources of 
electrical signals (or waveforms) which, when processed and 
passed through an appropriate amplifier/speaker system, 
become sound. We cannot, however, exclude tiie electronic 
processing of acoustic sound sources. Picking up sound 
with a microphone and passing it through an amplifier/ 
speaker system is, tef^ntcally speaking, electrorrio processing. 
For this reason it sometimes becomes quite difficult to draw 
the line between electronic music and non-electronic music, 
especially since electronic music often uses acoustic sound 
sources ~ sometimes with no alteration of the original sound 
quality. 

In this chapterwe shall look very briefly at some of the forms 
electronic music takes and explore methods for synthesizing 
sounds on a synthesizer. 

2-2 Live Electronic Music 

One form of live electronic musie invdvet ^e use of a un- 
recorded tape of electronically processed or generated sounds 
which is played on stage where one or more "live" solo 
instruments play in counterpoint to the tape. Another form 
of live electronic music involves the use of one or more 
synthesizers; the smaller stage type or sometimes the larger 
studio type. This type of music may or may not tneludftHip 
use of a pre-recorded tape. Acoustic instruments may also be 
involved. It is common to use human voices in their natural 
form and/or processed electronically. Stage acticm may be 
a part of the performance, as well as special lighting effects, 
including the use of lasers. 

2-3 Musique Concrete 

In musique concrete, portions of tape recorded sounds are 
cut up and patched together in different ways to form the 
finished composition. Many times the portions are altered 
before editing into the final composition. Among the many 
changes possible are abrupt stopping and starting of a sound, 
drastic or subtle changes in speed, playing the sound back- 
wards, filtering, or any combination of these. Any sound 
which can be recorded on tape becomes a candidate for this 
art form. Considered from this point Of view a taipe recorder 
can be thought of as a musical instrument Wfi^ as many 
creative possibilities as any other musical Instrument. 


14 Electronic Music 


Although the subject of musique concrete is not delt with in 
any detail in this book, its importance cannot be overlooked, 
in the eiatly days of electronic music, musique concri^ was 
considered to be a completely different form of music with 
no relation to music generated with electronic oscillators, but 
today this is not so. Indeed, musique concrete techniques 
often form an integrated part of synthesizer generated 
electronic music, and synthesizers are often used as sound 
sources and sound modifiers in musique cwwriite. 


2-4 Recorded Electronic Music 

In most forms of electronic music the final product Is a 
recording. In almtwt alt of these cases It vroutd be impossible 

to hear this recorded music in any other form. This type of 
electronic music usually involves tiie use of a multichannel 
tape recorder or the use of ordtrtary taps recorders in a 

fashion which imitates multichannel recording techniques. 
Usually, each voice in the composition is synthesized and 
recorded one at a time; This is very much like the process 

involved if we tried to record a symphoney by having each 
musicians come into the studio one at a time to record his/her 
part 

2-5 Waveforms 

The tuning fork is an example of a sound source vMldh 
vibrates only as a whole; therefore, it produces only the 
fundamental and no other harmonics. This sound is very 
clean and pure. If we could watch the movement of one of 
the prongs and graph this movement, we would get a diagram 
like that shown in Fig. 2-1 (a). The vibrating prongs of the 
tuning fork cause minute changes in air pressure, tf these air 
pressure changes were diagrammed, they would appear as 
shown in (b). tf we placed a microphone near the vibrating 
tuning fork, it would generate minute variations of electrical 
current whose voltage changes would be as shown in (c). 
An amplifier could be connected to the microphone and used 
to drive a speaker* The moving cone of the speaker would 
produce changes In air pressure which, assuming perfect 
equipment, would exactly mirror the air pressure changes 
origtnalty produced by the tuning fork. The only difference 
would be that air pressure changes produced by the speaker 
could be larger (amplified) or smaller (attenuated) than the 
air pr^ure change reaching the microphone, depending on 
the position of the amplifier volume control. 

The three diagrams shown in Fig. 2-1 represent the waveform 
of the sound produced by the tuning fork. The only dif- 
ference betvifeen these waves Is the medium; in ^er words 
(a) is the movement of a prong, (b) is air pressure changes, 
and (c) is voltage changes. In electronic music, of course, we 
are usually more concerned with ihe medium of voltage 
change. 

The waveform produced by the tuning fork is called a sine 
wave. Imagine a pen attained to the end of a pendulum. If 
we tAae» a ler^ of pt0st under tiie pendulum in such a way 
that the pen remains in contact with the paper and if we pull 


Fi«.2.1 WBVSfbrms 

® Movement of one prong 


Pitch = 440 Hz A 


Left 
Rest 

Right 


or 

one wavelength 


Appro x"*^ — 

,0023 

440 CyclB 

s = 1 Second 


Time for _ 1 Second 
one cycle Frequency 


® Air pressure changes as a result of tuning fork vibration 
+ Pressure 


Normal 
air 

pressure 






—JQ073 


Mcond 


— Pressure 


Q Voltage generated by microphone (fricklng up tuning fork) 
+ Voltage 


0 

Voltage 




■ .0023 



second 


Fig. 2-2 Wavefomrts in an Audio System 


Prong Air 

movement pressure Voltage 



Cone Air 
Voltage movement pressure 


AmpVtfieT 


9 000 O 



Electrantc Music 15 


the paper at a steady rate of speed as the pendulum swings 
the pen will draw a perfect sine wave, as shown in Fig. 2-3 
Sine waves are mathematically very closely related t^ the 
circle and they are very easy to generate using mechanical 
means that are based on the principle of the wheel. Ordinary 
hons^old elei^iGfty Is generated wJth machines which spin 
and thus produce current in the form of a sine wave Most 
courttrles use either 60 Hz or 50 Hz for household electricity. 
60 Hz produces the apprcwimate pitch of B below the bass 
cleft, while 50 Hz produces the approximate pitch of the A 
tJbelow that. If we had tuning forks which could produqe 
these pitches, the only difference between the sound of these 
tuning forks and the sound which could be produced by 
an ordinary house current would be one of intensity The 
sound of the tuning fork would also gradually die away. 

Since the complex vibration modes of different types of 
sound sources produce sounds with a different harmonic 
content, each type of sound source wfll produce Its own 
unique waveform. A few waveforms are shown in Fig. 2-4 


Fig. 2-4 Waveforms 
© Flute 


Pitch = D4 



© aarlrwt 

Pitch = G3 



Pitch = D4 



© Trumpet 


Pitch - F4 




Spectrum Diagrams 
Pitch = D4 


-40 




















































1 



2 


3 


* 

•I- 

r'l- 



Pitch = G3 


Frequancy 


3K 


dB 


-30 


-40 1 


100 

Pitch = F4 


300 


500 11 

Frequency 


4° I 


3K 


-30 


-40 1 







1 




















































i 




1 




2 




4 

S, 

B 

7 

a 









Frequency 


16 Electronic Music 


With each waveform is shown a spectrum diagram. Spectrum 
diagrams show the harmonic content of a sound or waveform. 
Frequency is shown atong the bottom of each diagram. Each 
heavy vertical line represents a harmonic and the length of 
each line represents the relative amount of that harmonic 
which is contained in flie tof&t sound. In synthesizing sotiiids, 
spectrum diagrams will often prove much more useful than 
drawings showing waveforms. Waveforms are sometimes 
helpful, however, as seen with the flute waveform which is 
close to a sine wave indicating that the flute tone quality 
contains relatively few, low-intensity harmonics. 

Additive and Subtractive SyifHiesis 

In electronic music there are two basic approaches to synthe- 
sizing sounds: additive synthesis and subtractive synthesis/' 

In additfve synthesis we start with sine waves of various 
frequencies and add them togatfier in the correct amounts to 
produce sound with the desired harmonic content. This 
method usually requires a large number of sine wave sources 
(one for each harmonic desired). The harmonic content of 
many sounds changes during its production, thus the 
accurate control of the amount of each indhfidual sine wave 
becomes quite compfex. For these reasons additive synthesis 
is not as common as subtractive synthesis, hut oven so, it 
is sometimes used to a certain extent with synthesizers 
designed for subtractive synthesis. 

In subtractive synthesis we start with a waveform which is 
already rich in harmonics and use electronic filters to remove 
unwanted harmonics to produce the desired sound. In place 
of the vibrating air column, string, mat&l plate, etc., we 
use an oscillator (electronic waveform generator) which is set 
to generate a waveform rich in harmonics. The output of 
the oscillator is connected to a filter which alters the basic 
tone coloring in ways not unlike the effects of the body 
of a musical instrument. In many cases, one filter is enough 
to easily synthesize at least passable imitations of many 
sounds, but it is usually advantageous to use several fitters to 
help imitate more closely the physical characteristics of the 
body of the instrument and the changes which occur during 
the production of each note. The oscillator, then, would 
be analogous to the vibrating portion of an instrument and 
the filter analogous to the body of the instrument. 

In subtractive synthesis, two of the more common waveforms 
are the sawtooth (or ramp) wave and the square wave, shown 
in Fig. 2-5. A glance at the mechanical symetry of these 
waveforms will hint at the fact that they are very simple to 
generate electronically. For example, a square wave can be 
generated very simply with a circuit which continually turns 
itself on and off. It is also easy to accurately control the 
frequency of such wave generators, this being an important 
consideration for accurate pitch control in music. The 
exactness of these waveforms also gives a hint of the regularity 
of the harmonic content of each, as shown in the respective 
spectrum diagrams. 

1. This discussion diitfesBfds cSract synthesis by computer which is 
presented later in tliis diapter (2-1 1 ; p. 23). 


Electronic Music 17 


Fig. 2-5 SYNTHESIZER WAVEFORMS 


HARMONIC SPECTRUM 
OF PERFECT SAWTOOTH 
WAVE 



-10 


-20 


dB -30 


-40 


-60 


-60 

HARMONIC NUMBER i 
FREQUENCY x 


2 
2X 


3X 


4 
4X 


5 

5X 


6 7 8 9 10 
6X 7X 8X 9X IIK 


HARMONIC SPECTRUM 
OF PERFECT SQUARE 
WAVE 


0 

-10 
-20 
dB -30 
-40 
-50 
-60 

HARMONIC 
FREQUENCY 


3 

3X 


5 7 9 II 13 IS 17 19 
5X TX 91C II X 19X 


18 Electronic iVlusic 


In Fig. 2-6 (a), three sine waves representinn the frequencies 
X, 2x, and 3x (in other words, the first, second, and third 
harmonics) are added together, the result of which looks 
much like a sawtooth wave. If more harmonics were added, 
the result would be a more perfect sawtooth wave. The 
perfect sawtooth wave (Fig. 2-5 (a)), #ien, contains harmonies 
which extend theoretically upwards to infinity. In Fig. 2-6 
(b), three sine waves of frequencies x, 3x, and 5x (odd 
numbered harmonics) are added together witfi the result 
looking much like a square wave. 

The sine wave and square wave are exactly symmetrical, both 
horizontally and vertically; the wave patterns continually 
repeat tfiennselves, and if they are inverted, i}tey look and 
sound the same, as shown in Fig. 2-7. When inverted, only 
the phase, or starting point, is different. Notice that if 
wavefonns (a) and (b) are added together, tiie reujit Is zero, 
or no sound. The same Is true of wa>»^rms (g| and (d). 

The sawtooth wave may seem vertically imsymmetrical 
because it looks different when inverted; the ramps go up 
instead of down. Fig. 2-8 {a) shows w^at happens when we 
add three inverted sine waves of frequencies x. 2x, and 2x. 
Since the sound of a sine wave does not change when it is 
inverted, we might expect that adding inverted sine waves 
together would not produce a sound different from that 
obtained by adding non-inverted sine waves, and this is 
exactly what happens. The sound of the inverted sawtooth 
wave and the sound of the non-inverted sawtooth wave are 
exactly the same. For this reason, it is not important whether 
a synthesizer produces upifoing or down-going ramps as a 
source for sawtooth waves. 

In Fig. 2-6, all harmonics are in phase with each other. In 
other words, at the point where the harmonics all cross the 
center "0" line at the same time (as they do at the beginning 
of the periods diagrammed), they are moving in the same 
direction: up. In Fig. 2-8 (a), they are all moving down. 
Fig. 2-8 (b) shows what happens when we try to make a 
square wave with odd numbered harmonics, but invert one 
(the third harmonic) so that it is out of phase with the others. 
"Oie important point here is that although the resulting 
waveform does not look anything like a square wave, it 
contains exactly the same harmonics as the wave in Fig. 2-6 
(b), thus it will sound exactly the same. The tone color 
of a sound source, then, will depend on the harmonic con- 
tent of that sound and will usually have no illation to the 
phase relationships of those harmonics except in sound 
sources where the phase relation is in constant change (an 
effect produced by a ^ase ^ifter}^ Thts is why spectrum 
diagrams are usually more useful ^an knowing the waveform 
of a sound source. 

Even so, in some cases knowing the waveform of a sound 
$oun» can be helpful, as with the Hute example cited in 2-5 
above. Another example is that of the clarinet, whose wave- 
forms are shown in Fig. 2-4 (b). These waveforms are clearly 
quite near a siquare wiave and diej^ir^ tiie spectfum diagram 
reveals that tiie odd numbered harmonics are stronger than 


Ffitr. 2-S Addttion of Hennonics 

Q 






Spectrum 


Spectrum 


ii 


1 2 3 
Perfect saMttooth 


1 2 3 4 5 
Perfect square wave 


Fig. 2-7 Inversion of Waveforms 

0 



Electronic Music 19 


the even numbered harmonics. This, and the shape of the 
waveforms, suggests that the square wave would be a good 
place to start in synthesizing a clarinet. This is borne out by 
the fact that the perfect square wave already sounds much 
like a clarinet without any processing. Note that the wave- 
forfns for ^e two pitches shown are a little different, indi- 
cating that ^e tone color of the instrument changes with 
pitch. 

The Voltage Controlled Synthesizer 

All voltage controlled synthesizers are basically alike^ Tfi* 
main differences are the package, the number of elements 
or modules available for synthesis, i^e number of communis 
tion points between the musician and the internal circuits, 
and the sophtsticstion of the circuits. The principals of syn- 
thesis remain the same for all synthesizers. 

Voltage control is based ott the eonc^Dt of modulation: the 

control of one parameter by another. Most sound sources 
produce sound which is quite complex. During the production 
of only one note, many acoustical events may take place. 
The most common are changes in loudness (the envelope) 
and changes in the harmonic content of the sound. Most of 
these events occur much too rapidly for them to be TOntrol- 
led manually. This is where voltage control comes in. If we 
know exactly how a parameter (such as loudness or harmonic 
content) will react to a $peciffc voltage change, it is often 
relatively simple to provide controlled voltage changes 
which will produce the exact effect desired. With systems 
which allow free interconnection of the synthesizer elements 
(patching), another advantage of the concept of voltage con- 
trol is that it is possible to connect any element output to 
any other element input, thus providing a tremendous 
variety of possibilities in synthesis. 

An Approech to Subtrectiva Synthesis 

The synthesis of sound is an art In itself. The most important 
ingredients to its mastery are patience and practice; practice 
particularly in the sense of familiarity with the synthesizer 
controls and their effect on the output sound. 

Fig. 2-9 shows the basic patch or connection of the three 
basic synthesizer element which are the most closely related 
to the three qualities of sound: pitch, tone color, and foud- 

ness. As shown, the arrangement is relatively useless because 
there are no control (modulation) inputs. These will be 
added in later chs^iters. 

The voltage controlled oscillator or VCO is the basic Source 

of pitch. An oscillator is simply an electronic circuit which 
generates electrical waveforms. The frequency or pitch of 
these waves is controlled by means of a control voltage. 

The most obvious source of pitch control voltage would be 

a keyboard, but there are many other possible sources. 


Fig. 2-8 rn\wrsion of Hsrmoitks 

(a) All inverted (b) Third harmonic inverted 

(Sawtooth ) (Square) 



-f- ■¥ 



spectrum Spectrum 



Fig. 2-9 The Basic Synthesizer Patch 


Pitch Tone color Loudniess 



Pitch Tone color Loudness 

control control control 

input input Input 


The voltage controlled filter or VCF is the basic element 
concerned wi:^ control of tone color. A filter is an electronic 
circuit which is able to remove or accent desired frequencies 


20 Electronic Music 


or harmonics in a sound source. The particular frequencies 
or harmonics acted on can be decided by means of a control 
voltage. The advantage of voltage controlled filter character- 
istics is with sounds in which the harmonic content changes 
during the production of a note and for creating the tone 
color ehangiBs in thoiiB sounds vMdh are pdrticulaHy asso- 
ciated wilti the synthesizer. 

The voltage controlled amplifier or VCA is used to control 
the articulation of the sound by means of a control voltage. 

Of the three qualities of sound, pitch and loudness usually 
present little problem in synthesis. If we want to synthesize 

a piccolo or a pizzicato string bass, it is extremely simple to 
decide and set the correct pitch range, and by repeatedly 
pressing a key on the Iceyboard while adjusting the loudness 
related controls, we can easily arrive at the correct envelope 
for the desired sound. Tone color is a different matter, 
however, and often requires much trial and error no matter 
how logically it is approached. This is where practice and 
patience will pay off. Tone color is also strongly affected by 
pitch and envelope or loudness. For^ese reasons, it is usually 
best to approach the synthesis of a specific sound by first 
deciding and setting the correct pitch range, then the correct 
loudness contour or envelope. Once these qualities of sound 
are more or less correct, it is possible to approach and 
experiment with tone color. 

2-9 Noise 

The VCO forms the basic SOWld source for synthesizing 
pitched sounds. Most syntfiesizers also include another 
source of sound called the noif generator whidi Is used for 
synthesizing non-pitched sounds. Usually, two types of 
noise are provided: white noise and pink noise. White noise is 
the random combination of all frequencies and produces 
a hissing sound much like that which can be heard when an 
FM radio tuner is set between stations. Pink noise contains 
fewer higher frequencies and produces a rushing Sound much 
like a waterfall. Noise is discussed in a little more detail in 
Chapter 5 (p. 51 ). Fig. 2-10 shows a noise waveform. 

2-10 Computer Music 

The computer is becoming more and more common in the 
production of electronic music. Many of the fi rst experi ments 
involving computers and music were aimed at trying to have 
computers "compose" music, thus the term computer music 
may often be much misunderstood. This approach is no 
longer common. 

The use of computers in music more recently (an be divided 

into two basic categories: 

1. The use of computers to completely or partially con- 
trol a synthe${:»r. 

2. Dir^ syntiiesis in which a computer is used to generate 
any desired wave^nm. 


Fig. 2-10 Noise Waveform 



Before discussing these two categories, it might be well to 


Electronic Music 21 


look at the question of why musicians would want to use a 

computer in music. 

At best, the score is only a poor representation of the musical 
sounds the composer/arranger intended, and, therefore, 
must be translated or interpreted. Fig. 2-11 illustrates this 
process. 


Fig. 2-1 1 Translation of Score into Sound 


r 


SYNTHE- 
SIZER 


COMPUTER 


INSTRU- 
MENT 


PLAYER 


MUSIC (SOUND) 


X 


INSTRU- 
MENT 


INSTRU- 
MENT 


PLAYER 



INSTRU- 
MENT 



CONDUCTOR 


COMPUTER 

(DIRECT 

SYNTHesiSI 


SCORE 


COMPOSER/ASftANGER 


Normally, the conductor interprets the score. In one passage 
the horns should be louder than the other instruments, and 
in another the music should slow down with these notes being 
held a little longer than the other notes, and so on. During 
the actual performance, the conductor can normally use only 
body motions to remind the players of what is desired of 
them. Even in the best of bands and orchestras there will 
be some loss as the music filters down through each player to 
the instruments, and thus finally emerges as sound. Using 
a computer, the composer/arranger may translate music into 
sound without the need to filter it through a group of 
people. 

It is a simple matter to translate a score directly Into com- 
puter data for the production of music, but the result 
would be very mechanical sounding. The result would be the 
same if a conduetor conducted the score without trying to 
interpret it in any way, except that the computer could 
produce more accurate timing. It is very important for the 
musician who uses a computer to interpret the music and 
incorporate these interpretations into the computer program. 

Returning to the two categories in which computers are 
used, in the first category a computer is programmed with 


22 Electronic Music 


numbiers whf€^ represem ^ voltage lev^ pul$e -Uitiings 
desired by the musician for the control of an ordinary 
voltage controlled synthesizer. The computer, in effiBct, takes 
the place of the keylx>ard controller or other controllers used 
vvith tiie synthesizer. 

Most electronic music is recorded one melody line at a time, 
hence it is very difficult to judge how the different parts 
will sound when mixed together. The result is often unac- 
ceptable and some or all of the parts must be recorded over. 
Computers offer a large advantage in this area since they can 
be pFognenmed to provide simultaneous control of several 
melody lines thus allowing ^e musician to hear how the 
combination will sound. 

Within the category of computer controlled synthesizers are 
two approaches. The first is to use a standard computer 
system in which a special "music" program would hgve to 
be written and implemented. This requires an intimate 
knowledge of computer programming which becomes one 
of the major drawbacks of this approach. The other major 
drawback is that it would usually be necessary to design and 
build custom interfacing units in order to convert the com- 
puter outputs into control voltages and gate pulses which the 
synthesizer can use. If the musician is not interested in 
learning enough to do this, he/she might be lucky enough to 
enlist the aid of a computer hobbyist. The larger of the home 
computer systems is capable Cif hendling a rather sophisticated 
music program which would make it quite easy for the 
musician to use. The smaller of these systems would also be 
capable of music programs, but would be rather tlniitad In 
what they could do and would probably require a rather 
specialized music encoding procedure. 

The second approach to computer controlled synthesizers 
Would be fo use a specialized computer which hes been 
designed so that its sole purpose is the control of a synthesizer. 
The Roland IVlC-4 MicroComposer is the first example of 
this type bf computer to be put on the market It uses a 
simple encoding system which was first developed by com- 
poser Ralph Dyck for use with a programmable digital 
sequencer which he built for his commercial studio in 
Vancouver, Canada. 

The fundamental difference between the MicroComposer and 
a digital sequencer, programmable or otherwise, is that the 
MicroComposer was designed around an integrated Circuit 

called a microprocessor or CPU (Central Processing Unit). 
It is the microprocessor which is responsible for the versatality 
of the MicroComposer. Since the MicroComposer was 
designed with the musician in mind, it requires no knowl- 
edge of computers or computer programming. It pro- 
vides control for as many as eight melody lines plus as 
many as six percussion voices, all completely independent 
of each other. Its operation is very much like that of an 
electronic desk calculator and, due to the use of a micro- 
processor, the MicroComposer provides a great number of 
operating functions which give the musician an unprecedented 
amount of contral over the music betng produced. To produce 
a similar amount of cormol in a home type computer would 


Electronic Music 23 


require a rather large system and sophisticated programming 
techniques. 

2-11 Direct Synthesis 

In the second category of computers used In music, a series 
of numbers whii^ represent the instantaneous values of 
voltage levels at different points fh a waveform are stored In 
a computer memory and called forth as needed to produce 

sound. This is direct synthesis. 

It is easier to understand direct synthesis by first demon- 
strating how a waveform can be analyzed and stored in a 
computer as data. This is done by sampling the sound wave 
at regular closely spaced intervals. Each sample would 
represent the instantaneous voltage level of the sound wave at 
the sampled time. Each of these voltage values is converted 
Into a number which is then stored in its own space in the 
memory. This Is shown In Fig. 2-12 (a). 


Fig. 2-12 Direct Synthesis 
® Wave sampling 

WAVE: 


VOLTAGE LEVELS ATSAMPLED TIMES: 


+10- 


+5- 


Level 0 


-5- 


-10- 


MEMORY 
DATA: 


Ti=0 

Ta -4^ 
T3 =+6iO 
T4 = 47.0 
Ts = +8.7 
Tfi = +9.7 
T7-+10.0 
Tg=+9.7 



+10- 


+5- 


-5- 


-10- 


15 20 


10 


25 


(b) Wave generation 
VOLTAGE LEVELS: 


+10- 


+5- 


STORED 
DATA: 


T, =0 
T2 = +2.6 
T3 - +5.0 
T4 *+7.0 

Ts = +8.7 
T,, = +9.7 

= +10.0 
Tg = +9.7 


etc. 


WAVEFORM: 


-10- 



+10- 


+5- 


-10- 



ete. 


24 Electronic Music 


To retreive the sound wave, the computer reads the stored 
data in the memory much as a person might run a finger 
steadily down a column of figures. As each number is read, 
it is converted into its corresponding voltage level. Each 
voltage level is held until a new level comes along. The total 
result Is a sound wa^ whfdi Is squared off; It has only hori- 
zontal and vertical lines, no slants or curves. A filter is used 
to smooth these corners off. This process is shown in Fig, 
2-12 (b). 

First, It can be seen that the sannpling rate would have to 
be rather fast compared to the sound wave. The more samples 
taken, the more closely the reproduced sound wave will 
resemtile the original. There is a definite upper limit of 
sampling speed which each computer can handle. 

Second, there will be a limitation to the numbers available 
for showing voltage levels. In the example shown In Fig. 2-12, 
if the computer cannot haindle the digits to the right of the 
cecimal point, the data for the sine wave would have to be 
rounded off and written: 



= 0 

Tft =10 

Ta 

= 3 

T, =10 

Ta 

= 5 

T„ =10 

T4 

= 7 

Ts =9 

Ts 

= 9 

T,o =7 
ETC. 


Thus it will be less like the original sine wave than the 

original example was, as shown in fig. 2-13. 

Since numbers representing the waveform are stored in the 
computer memory, it becomes a very simple matter to 
alter the wave by altering the numbers. From this it can be 
seen that it is possible to invent any waveform and convert 
it to data for loading into the memory. This is direct synthesis 
which can exactly imitate any known sound, or can create 
completely new sounds. 

The major disadvantage of direct synthesis is that it requires 
an extremdly targe computer system and Is thus limited to 

large institutions. It aiSO requires an intimate knowledge 
of computers and programming, including advanced mathe- 
matics. 


Fig. 2-1 3 Dfstprtlon in Dirw^ Synthesis 


+10- 



-10- 


+10- 


+ 5- 


-10- 


Major 
□Utortlon 



Electronic Music 25 


2-12 Questions 

1 . List sorne of the forms which electronic music can take. 

2. Draw a sine wave. Explain this drawing in terms of air pr^sure changes. In terms of variations of electrical 

current. 

3. Explain additive and subtractive synthesis. Which is more common? Why? 

4. What are two commonly used waveforms in subtractive synthesis? Why are they good for subtractive synthe- 
sis? Give the harmonic content of each. 

5. What effect does inverting a waveform have on its tone color? What effect does inverting some of the har- 
monics in a sound have on ^e waveform? On the tone color? 

6. What is the advantage of voltage control in a synthesizer? 

7. In vi^at ways are a stage type synthesizer and a large studio system synthesizer different? In what ways are 
they alike? 

8. What are the three qualities of sound and what are the synthesizer elements most associated with eac^? 

9. Generally, in which order should thethree qualities of sound be considered when synthesizing a sound? Why? 
10. What are the two categories in which computers are used in electronic music? 


Words to define: 

additive synthesis 
computer music 
direct synthesis 
electronic music 

modulation 
musique concrete 
ndse 

noise generator 

oscillator 

phase 

pink noise 

sawtooth wave 

sine wave 

spectrum 

square wave 

subtractive synthesis 

synthesis 

VGA 

VCF 

vco 

voltage controlled amplifier 
voltage controlled filter 
voltage controlled oscillator 
voltage controlled synthesizer 
waveform 
white noise 


Pitch 


Chapter Three: 


3-1 Introduction 

The first quality of sound which we shall discuss in detail is 
pitch. Pitch is that quality of sound in which some sounds 
seem higher or lower than others. In the voltage controlled 
synthesizer, the mainsource of pitch is the Voltage Controlled 
Oscillator or VCO. 

One of the most common sources of control voltage for tfie 

VCO is the keyboard controller. The keyboard controller 
is nothing more than a voltage divider; the level of the voltage 
output depends on which key is pressed. When the control 

voltage is applied to a properly calibrated VCO, the VCO 
will generate a pitch which is related to the key pressed. 

In order to better understand the relation between the 
frequency output of the VCO to tfie voltage input, it is 
necessary to review the construction of our musical scale 
system. 


3-2 Pitch Relationships in Music 

In music, instead of frequencies, pitches are given letter 
names or syllable names, or sometimes numbers. 











) ^ „ o ■■ ° 




CDEFGABC 
DO RE Ml FA ^ LA Tl DO 

1 2 3 4 5 6 7 8 (or 1») 

When more than one note is sounded together, this is called a 
diofd.'' The tssult we hear as harmony. Different chords 
produce harmony which to our eai% has different qualities. 

The distance between two pitches is called an interval. The 
two qualities of an interval are: consonance and dissonance. 
A consonant interval (sometimes called a concord) is an 
interval which to our ears seems to be harmonically at rest 
or comfortable. A dissonant interval (sometimes called a 
discord) is an Interval which seems jarring to our ears and not 
harmonically restful. 

It is possible to play a number of intervals and have a group 
of people judge in which order they shcnild be placed so that 
u^en flayed they would move gradually from consonant 

to disson^ant. Somewhere near the middle of this progression 
would be a dividing line between consonance and dissonance. 
The postfton of thfs dividitig line has changed over the years 

so that chords which we consider consonant now would 
have been considered dissonant in the earlier periods of 
muiic. 

The interval which causes the least amount of jarring when 

it Is heard is unison: two pitches of exactly the SEime fre- 
quency. The next most pleasing interval is the octave. In the 

1. Some scholars would prefer to define a chord as having at least 
three different pitches, but this point is not important to the 
pr^ent discussicm. 


28 Pitch 


3-3 


octave, the frequency of the upper pitch is exactly twice that 
of the lower pitch. Most musical scale systems are based on 
the relatlonshfp of the octave. For example, the Arabs use an 

octave divided into equal sixteenths, and the Hindus divide 
the octave into twenty-two steps, but use only seven of these. 

The Western system of music usually divides the octave into 
twelve parts. In the early days of music, these divisions were 
tuned in what is called just (as in "right") intonation. Although 
the twelve divisions are unequal, the frequencies of these 
pitches have very dose relations toeacho^r. Asan example, 
the perfect fifth has a ratio of 3:2. In other words, If the 
upper pitch is 300 Hz, then the lower pitch will be 200 Hz. 
Fig. 3-1 shows the frequenby ratios of ^e Intervals in the |ust 
tuned scale. It should be noted that the consonance or 
dissonance of an interval Is closely related to the ratio of the 
frequencies of that interval. The smaller the numbers used to 
express the ratio, the more consonant the interval. 

The scale of just intonation produces harmony which is 
pleasing to the ear because of these interval relations, but it 
causes major problems with instruments using fixed tuning 
systems such as the piano or organ keyboard and guitar 
frets, tt is impossible for this type of instrument to modulate 
to distantly related keys without retuning the instrument 

The linabtHtv to modulate freely was one of the reasons for 

the development of what we call the scale of even (or equal) 
temperament. In this scale, the octave is divided into twelve 
agua/ parts. The frequency ratio between any two adjacent 
notes (semitones) is exactly the same. With equally spaced 
pitches it becomes obvious that the intervals in a scale 
ratalti the tame mtadohshfp no matter which note Is used 
for the starting point. The frequency ratio of a major sixth 
will remain exactly the same no matter which pitch is used to 
build it on. The unaqinil dNtlons of the Just scale will form 
correct intervals only wrtien certain pitches are used as the 
starting point. 

Fig. 3-2 shows the frequency ratios for the scale of even 
temperament. Fig. 3^ shows a comparison of the frequencies 

between the scale systems using a one octave scale starting 
on middle C. If we assign middle Cthe frequency of 264 Hz 
(which It often is In setentific iaborsitoriei), the frequents 
of the two scale systems would be as shown. Also note that 
expressing the frequencies of the just scale accurately requires 
fewer digits. 

Beat Frequencies 

AltfiouE^ Western music primarily uses the equally tempered 
scale, the scale of just Intonation is not obsolete. The violin, 
for example, has no frets and can, therefore, play intervals 
of any frequency ratio. Most good violinists have a tendancy 
to sound just intervals in whatever key they are playing>eii!en 
when playing with inflexible evenly tempered instruments. 

The reason for this is the interaction which takes place 
u^en tira dcHdy related pitches are sounded together. 
This interaction tikes the form of a wavering in the loudness 


Fig. 3-1 Frequency Ratios for Scale of Just Intonation 
(a) Interval RatftA 

Interval 


Frequency ratio 
from starting point 


Unison .,* 1:1 1 

Semitone 16:15 1.066667 

Minorlone 10:9 1.111111 

Major tone 9:8 1.125 

Minor third 6:5 1.2 

Major third 5:4 1.25 

Perfect fourth 4:3 1.333333 

Augmented fourth . . . 45:32 1.40625 

Diminished fifth 64:45 1.422222 

Perfect fifth 3:2 1.5 

Minor sixth 8:5 1.6 

Major sixth 5:3 1.686667 

Harmonic minor seventh .... 7:4 1.75 

Grave minor seventh, 16:9 1.777778 

Minor seventh 9:5 1.8 

Major seventh.^ i . . ...... . . 16:8 1.875 

Octave 2:1 2 


Cents* 
from start 
point 

0 

111.731 
182.404 
203.910 
315.641 
386.314 
498.045 
590.224 
609.777 
701 .955 
813.687 
884.359 
968.826 
996.091 
1,017.597 
1,088.269 
1,200.000 

'(See p. 32) 


(b) Intervals in Order from Consonance to Dissonance 


liMHwtt iSHlon btt&« 


Fraqumoy 


2:1 


3i2 


Major 
tIMh 

S:3 


U46r 
thtrd 

S:4 


4:3 


Minor 
thtid 


KB 


Minor 
itxth 

S:6 


Fig. 3-2 Frequertcy Ratloe for Equally Tempered Scats 


Frequency ratio 
" ^"'^ from starting point 

Unison 1:1 

Semitone or minor second . 1.059463:1 

Whole tone or major second 1 .1 22462:1 

Minorthird 1.189207:1 

Majorthird 1.259921:1 

Perfect fourth 1 .334840 : 1 

Augmeinid fourth » 1414214-1 
DIminUhed fifth f"-f 

Perfect fifth 1.498307:1 

Minor sixth. . . . ^ , 1 .587401 : 1 

Majortixth 1.681793:1 

Minor seventh 1.781797:1 

Major seventh 1.887749:1 

Octave 2:1 


Cents' from 
starting point 

0 
100 
200 
300 
400 
SCO 

600 

700 
800 

900 

1,000 
1,100 
1.2Q0 
•(See p. 32) 


Fig. 3-3 Comparison of Scale System Frequencies 

Just o o o o o q 

Intonation S R 9 f:i SS 2 


8 


Equal 
Tempering 


•5 


>- CO 

(0 n 
w ci 

ID 

m 


Ul u. (9 < m u 


S3 


of the total sound and is called a beat. Fig. 3-4 illustrates 
this. For simplicity, 6 Hz and 5 Hz are shown rather than 
audio frequend«n. Vt^ve^nti (c) mipresents the algebraic 
addition of waveforms (a) and (b). The changes in loudness 
can be seen clearly in (c) where the sound starts at maximum 
Idudn^, dies dbwh t& minimunri at the center of #i« #ar 
gram, then expands again to maximum loudness jfftlJWHld of 
the period shown. In other words, the frequency iatf liie 
loudness variations is one cycle per secomi, at 1 Hz. Ndte 
that 6 Hz — 5 Hz = 1 Hz. For near unison, then, the beat 
frequency will be the difference between the two pitches. 

Other intervals will also produce beats when they are mistuned. 
The ease with which these beats can be heard wilt depend on 
the consonance or dissonance of the interval; the more con- 
sonant, the easier they are to hear. 

As an example, if we start with a pitch of 440 Hz, a perfect 
fiftfi above this would be 660 Hz because the frequency ratio 

ofaperfect,fifthi83:2(-| = -||^). If t^p^r #itBh 

happens to be 663 Hz, then we will hear a beat frequency of 
3 Hz because the upper pitch Is 3 Hz higher than the cor- 
rect 660 Hz. 

In the equally tempered scale, the frequency of a perfect 
fifth above 440 Hz is 659.26 Hz (see chart facing p. 66). 
The result is that when an equally tempered fifth is sounded, 
we hear a beat of 0:74 Hz (660 - 659.26 = 0.74). A good 
violinist would have a tendancy to adjust his/her pitch to 
eliminate this beat when playing harmony with other Instru- 
ments. 

Musical instruments are very often tuned using beat fre- 
quencies. The instrument being tuned is first adjUlted so that 
it produces a pitch at approximate unison (or some other 
interval, if desired) with a reference such as another instru- 
ment, a tuning fork, or a test oscillator. The instrument is 
then sounded together with the test pitch and fine tuned to 
eliminate ^e best 

The presence of beat-producing intervals in the equally 
tempered scale is a useful point to keep in mind when 
working with electronic music. Many sounds in electronic 
music are of a character which makes these beats easier to 
hear than if the music were played on acoustic instruments; 
therefore, it is sometimes necessary to compensate for this 
by hiding tflese beats under the oversll musical texture, or 
by employing some other means to correct offending pitches. 

The Natural Harmonic Series 

In Chalrter One it was shown how vibratory systems tend to 
Ibe quite complex, vibrating at various frequencies at the 
same time. These frequencies are called overtones. In most 
systems, these overtones have definite mathematical relation- 
ships with each other so the upper overtones are all multiples 
of the lowest or fundamental. Overtones of this type are 
harmonies. 


Fifl. 34 frequency WavB^nfl 



30 Pitch 


If we design an open end pipe or a monochord which pro- 
duces a fundamental at the pitch of A2 {bottom space bass 
cleft), and If wd ucfi'^tt mcisleal notate^^^Bm tasNtw^»' 
pitches of the possible harmoni^j th9 result up to the 
sixteenth harmonic would be as stiolvn in Fig. 3-5. This 
diagram gives a compartfbh of tfte actual fraquencies of the 
harmonics in relation to the frequencies of the equivalent 
pitches in both the scale of just Intonation and the equally 
tempered scale. Note that the hamitinfes ihdwn as ^aattar 
notes do not match the frequencies in either scale system. 
Also note that except for these quarter notes, the pitches in 
the scale of just intonatfon m&Ai tbe harmonics. And last, 
that in the tempered scale, CHlly ^^..OE^ves of the funda- 
mental {all the A's in this case) mate*i the harmonics. This 
series of pitches is called lhe natural hannanlc s«1es. 


FiB. 3^ Natural Harmonic Series for A2 
A 


gWt 


3r 


0 <> 


Nbttib of nets: A3 

A3 

E4 

A4 


Es 


As 




EiS 


G« 

0#« 

A« 

Nunttierof hamionio: 1 

2 

3 

4 

s 

6 

7 

8 

9 

10 

11 

12 

13 

U 

15 

16 

Frequency of 


u 


w 


-0 

00 

(O 


M 

CO 



?> 


true harmonic: 3 

w 
0 

u 
0 


oi 
0 

s 

•J 
0 

s 

(O 

0 

0 
0 

0 


to 
0 


3 

s 


Actual frequencies of the notes shown: 

u 


Juit 
intonation: 


Equel 

tempering: 


g 


o o 


(Jl 

s 

o 


00 to 

S 8 

b b 


3-5 Consonance and Dissonance 

With the preceding as background it is possible to give a 
better understanding of what causes consonance and dis- 
sonance in musical intervals. In Fig. 3-6 are shown some 
pitches with some of their harmonics written above them. 
The pitch in (b) is shown one octave above the root in (a). 
Note that if these two pitches are sounded together, most of 
the harmonics In (a) reinforce the first harmonics in (b). 
"(^O'K that in (c), which shows the perfect fifth above the 
root, fewer harmonics match. With the strong dissonance 
of the minor second shown by {e), none of the harmonics 
match. Also note that (a), (b), (c), and {d) form a major 
triad and that when sounded toge^er many of the hannonrcs 
coincide with each other. 


\ 


Pitch 31 


Fig. 3^ 6eniw>an^ipKl DkiatBrK» 


® ROOT 


(b) OCTAVE 
(Consonant) 


© FIFTH 


©THIRD 


Harmonic number 



@ MINOR SECOND 
tDissonant) 


Whole notes rapretent musical pitch; quartnr notes 
nqsresem^ lowmr hamnonlcs of those pitthes. 


3-6 Pitch Standard 

Usually, the A above middle C is used as a standard for 
designating the frequencies of the pitches in music. The pitch 
of this A has changed over the years but most countries 
today use A = 440 Hz. Throughout history, however, this 
pitch has varied from about 373 Hz to about 462 Hz. As an 
example, the A in Handel's time was 426.6 Hz. 


3-7 Exponential Prograuiom 

The relation between pitch and frequency is an important 
one, especially from the point of view of electronic music. 
If we start at the bottom df the piano and play a chromatic 
scale all the way to the top of the keyboard, to our ears it 
will sound as if all the pitches move upward in evenly spaced 
steps. For example, ihe distance between e C end a D et the 
bottom of the keyboard and at the top of the keyboard will 
sound the same because the ratios between the frequencies 
are the same. The frequeneies themselves, however, are quite 
different. The difference between the lowest C and D is about 
4 Hz. The difference between the highest C and D is about 
256 Hz. 

If we start at the bottom of the keyboard and ptay all the 

A's movfr^ tilW^ntlt, the distance between each pair of A's 
remains musiieally ttie same (one octave), but the frequency 
doubles with ead) A. This kiN of progression is called an 
exponentid progression. 

In measuring things we usually use a linear scale In which 
the divisions are equally spaced. The divisions on our rulers 
are equally spaced. A centimetar on one end is just » long as 

a centimeter on the other end. The compass has 360 equally 
spaced degrees. A day has twenty-four evenly spaced hours. 

With some things it is more convenient and practical to use 
an ncponential sc^le. The ^}de rule is a good «eampie. 

The divisions of most of its scales are unequally spaced; a 
measure of one unit is longer on the left end than it is on 


the right end. Moving from left to ri^, the difftartees^r otie 

unit become progressively shorter. 

Fig. 3-7 shows a comparison between a linear scale and an 
exfKmential scale. The upper divis1or» repre^m oetave 
intervals. Frequencies are shown below this and result in 
an exponential scale, a scale in which the divisions become 
pra^«5Slwly smaller towards the right. Most graphs which 
deal with sound use an exponential scale to show frequencies 
because this has a "linear" sound to our ears. 


Fig.3-7 Pitch/Frequency Comparison 
PITCH (linear prosression): 


41 


Ai 


Ai 
—i— 


-Middle C 


A6 


FREOU£NCV toqWiMntial prograMfon): 


H — I 1 r I I I 
9 S S^SSS 


t ' r 1 I I i 


H — I — I — 1 I I n 

O O OOOOOQ 
O O OOOOQQ 
O O OOOOOO 


Cents 

In the above, the differences between the frequencies of the 
pitches of a mfrfor ^MM>hd at the top of the kieyboard and at 
the bottom of the keyboard show that the use of frequency 
for the measure of the accuracy of pitches is inconvenient. 
For this we use the cent. The cent is based on the equally 
tempered scale with 1200 cents equal to one octave. This 
means tfiat a minor second is 100 cents. In experiments it 
has been shovm tfiat when two independent pftches ate 
sounded one after the other for comparison, the average 
person can detect a difference of pitch as small as about 
three cents. 


Voltage to- Frequency Relations 

From the above, it can be shown that there are two possible 
practical approaches to establishir^ fttlationshlp betwran 
the amount of control voltage change needed to produce a 

given VCO pitch change. 

The first of these systems uses what is known as a linear 
VCO. Ih ^tk type VdO, the ft^sqaeney output of the VCO 

is directly proportional to the control voltage applied. For 
example, such a VCO might have the following vottage-to- 
frequency r^atlons: 

control VCO 
voltage frequency 
input output 

1 volt = 110 Hz (A2) 

2 = 220 (A3) 

3 = 330 (approximately E4I 

4 = 440 (A4» 
etc. 


Unfiar 
prf^Sitsion 


Ejqionential 


Or, to produce ^e octaves only: 


Pitch 33 


Exponential 

voltage 

progi^on 


control VCO 
voltage frequency 
input output 

1 volt = llOHzfAz) 

2 * ^ (A3) 
4 - 440 (A4) 
8 » 880 (A5) 

etc. 


Linear 
pitch 

progression 


Linear 
voltage 
progression 



Linear 
pitch 

progression 


To produce a series of octaves moving upwards, therefore, 
requires a control voltage source which incressss In exponen- 
tial steps. 

The second approach uses what is known as an exponential 
VCO. In this type VCO, the pitch output of the 
VCO is directly proportional to the control voltags apidied. 
For example, such a VCO might have the following voltage- 
to-frequency relations: 

VCO 
frequency 
output 

110 Hz (A2l 
220 (A3) 
440 (A4) 
880 (As) , 
etc. 

In this system. It is neeessary to provide linear changes In 
voltage to produce a progression of octaves. 

3^10 TheUnearVCO 

Fig. 3-8 (a) shows a plot of the voltage-to-frequency rdations 

in the linear VCO of the example in 3-9 above. From this it 
can be seen where the term "linear" originates. In (b) is 
shown the plot for the voltage-to-^tdh [relations for the same 
VCO. The resulting plot forms an exponential curve. To 
produce any even progression of pitches, such as octaves 
(as shown) or a chromatic scale, would require exponential 
changes in the control voltage source. This fact produces 
problems in connection with tuning and transposing. If we 
have a voltage source which produces an exponential sequence 
of voltages, as for example: 1v*, 2v, 4v, 8v, and feed this 
sequence to the linear VCO of this example, it will produce 
pitches which make one octave jumps starting at 1 10 Hz A2 
and ending on 880 Hz As, as shown in (b). To raise this pitch 
sequence one octave so that it starts on 220 Hz A3 and ends 
on 1760 Hz would require that we double (multiply by 
2) the Input control voltage. The linear VCO, then, needs the 
addition of an electronic multiplier at its input if it is to be 
able to transpose freely. It also requires an exponential 
keyboard, or a keyboard which produces voltage steps which 
increase exponentially when a chromatic scale is played. 
Since the keyboard control voltage is often used for other 
synthesiser control functions, other system parameters 
woufd also have to be deigned to ttitfteh Ihis exponential 
relation. 

3-1 1 The Exponential VCO 

Fig. 3^ fa) $hows a plotof thevottage-to frequency retadon 
in tfie exponential VCO. The main different between tiiis 

•tv = volt) 


Fig. 34 UnsarVCO 

® Voltage/Frequency ItoteSoni 

8 



200 400 600 800 IOOO 
Frequency 


(b) Voltage/Pitch Relation 



34 Pach 


curve and the one shown in Fig, 3-8 (b) is that it moves up 
and turns right instead of moving right and turning up. Fig. 
3-9 (b) shows the voltage-t<H)itch relation In ^ exponential 
VCO. From this it can be seen that the names which are 
applied to these two types of VCO depend merely on point 
df view. The exponential VCO produces a linear voltage-to- 
pitch relation and the linear VCO produces an exponential 
voltage-to-pitch relation. Electrical engineers are usually 
more concerned with oscillator frequency relations than with 
musical pitch relations, and it should be remembered that 
VCO's are used in other areas of electronics and not just as 
pitch sources for synthesizers. The names applied to VCO's 
are from the point of view of frequency. 

In music, since we are usually more concerned with pitch 
relations than frequency relations, the use of exponential 
VCO's where pitch is directly related to control voltage might 
seem the more logical choice for synthesizer pitch generation. 
Any linear source may be used to produce accurate musical 
scales^ and linear voltage sources are easier to match to other 
system parameters. Transposing and tuning become extremely 
simple, merely the mathematical addition of voltage sources. 
To transpose or tune an exponential VCO it is necessary only 
to supply a fixed voltage of the proper level in addition to 
the pitch control voltage. 

Slnea the exponential VCO uses Hnear inputs to produce 
exponential changes in frequency, it requires a linear-to- 
exponential converter. This exponential generator, as it is 

called, is an integral part of all exponential VCO's. Elec- 
tronically, exponential generatoi^ are not too difficult to 
design. The major difficulty is thattiiey are quite temperature 
sensitive, in earlier exponential VCO's, this dependency on 
temperature changes meant that the VCO's had to be con- 
tinually retuned since they would quickly drift off pitch. 
Temperature compensation used in modern VCO's, however, 
produces pitches vtrhidi are very stable. 

The exponential VCO is the most common VCO in use in 
voltage controlled synthesizers today. The most common 
voltage-to-pitch standard In use (s 1v/8raH volt per octave), 
as shown in the previous examples. This means that if the 
control voltage input is changed by one volt, the VCO 
pitch will change by one octave. To produce a pitch change 
of a semi tone would require a 1/12 volt change in control 
voltage since there are twelve semi-tones In one octave. The 
relationshTp of Iv/Bva Is qrulte convenTent. ft prababty eame 
about as the result of the design requirements of synthesizer 
circuits. With the most common power supply voltages used 
In many circuits eorrtaining IG^ Ontsgrated efreuift},, la 
relatively simple to obtain variations in^ol^^ Whl^mtiain 
perfectiy linear between 0 and apprOidmattety +12 volts. 
Witii- a ^^nstaid of 1v/8va and using +10 volts as an ap- 
prcxicYfni^' iUpper limit will produce a pitch range of ten 
octaves vt^tchi eovers all audible frequencies. If we specify a 
voltage ae^iracY of ^iO. millivolts (tCM)1 voltK tills will 

Fig. 3^10 shows tiie block diagram for a typical exponential 


Fig,3^ Ea^onatitte) VCO 

0 Voltaea^re^itieney Flelation 



Pitch 35 


voltage controlled oscillator. At the bottom of the diagram 
are three control voltage (CV) inputs. The KEY CV (from 
the keyboard) has an ONASFf= switch so that, if desired, the 
VCO frequency output will have tlo relation to the keyboard. 
The VCO in this example provides two other control voltage 
inpan, both VS^ voTuhie controls so that the input may be 
attenuated (reduced). The VCO TUNING control is nothing 
more than a variable voltage source Inside the VCO. The 
summing ampliffef does exactly wfhat the name implies: it 
adds all the control voltage inputs together to produce one 
voltage level to produce the desired VCO pitch. For example, 
if the teuF eotmol iroHage inputs were +1.50v, -0.82v, 
+1.32v, and %0Ov rwpectively, the sum would be: +2.00v. 


Fig. 3-10 VCO Block Diagram 


EXPONENTIAL 
VOLTAGE 
CONTROLLED 
OSCILLATOR 




Exponential 


Voltage 

Controlled 

Osdltator 

+ 

^ 





Modulation 

Deptli 

Contreto 




OUT 


OUT 


OUT 


OUT 


A A 


Key CV CV 
CV m IN 
IN 


•NOTE : Many VCO's do not prMde tine wave shapers since 
tbi^maa is^emlfy produced by filtsring the 
tifan^ wave. 


36 Pitch 


This voltage might produce the piteh of mtiE^e C, as an 
example. The actual pitch wilt depend on the design of the 
VCO but the standard used is not too Important because we 
are usually more eoti^ed >ui#i ^eresultins pMtNut tfw 
aefefit ifttemel voltw rwcass«v 10 pfoduds mt piteh. 

After the summing amplifier is the exponential converter 
which converts linear changes of voltage input Into the 
exponmttal changes wAtdi control the oscillator Itself. The 
most common form of oscillator is a type which generates a 
sawtooth wave; if other waveforms are desired, wave shaping 
drcuite must be Incorporated at the output as ^own In tiie 
diagram. 

The Low Frequency Oscttlator 

A low frequency oscillator or LFO is an oscillator which 
produces low frequencies usually starting from somewhere 
just above the lowest audio frequency (25 Hz or 30 Hz, for 
example) to frequencies far below the range of hearing. A 
typical LFO might reach as low as 0.01 Hz, which requires 
ten seconds to prfiduce one complete cycle. In some systems, 
VCO's can be used as LFO's, but most systems also provide one 
or more oscillators which specialize in generating low fre- 
quencies. In large systems, even the LFO's are voltaQe con- 
trollable with a voltage-to-frequency relation the same as the 
system's VCO's. Also, like VCO's, the LFO often produces 
several Wav&forms, the most common being the sine wave. 
If a low frequency sine wave is applied to a VCO control 
voltage input (Fig. 3-1 1 ), the result will be a wavering of the 
VCO frequency (pitch) at the UrO rtfte. This produces an 
effect whi^ in music is called vibrMp. 

Frequency Modulation 

Modulation teftirs to di« control of one parameter by 9nq#)^, 

the basic conc^^ Of the voltage controlled synllwslaser. 
Controlling the Trtquancy output by means of an «(tsmal 
control is a fomn of fteflKiency modiilirtiion (FM). Both 

vibrato and pitch control are forms of frequency modulation. 
Fig. 3-12 shows the waveforms obtained with LFO control of 
thff Vt!0. In (to) is shown *e VCO output vrithout modula- 
tion. In (c), (d), and (e) are shown the VCO output with 
various depths of modulation. The + and — excursions of the 
LFO viravefoitn in (a1 cause t^speetTva fnoreaMS and decreases 
in the frequency output of the VCO waveform. The deeper 
the modulation, the farther up and down the VCO frequency 
sweeps. The average frequency, or We frequency in the 
center of the VCO sweeps, is the same as the frequency 
would be without LFO modulation. 

Fig. 3-13 shows a block diagram for FM radio broadcasting 
which works on exactly tiie same prindpal. 

The Basic Synthesizer Patch 

Fig. 3-14 shows the basic synthesizer patch with the addition 
of VO^ modulate imm- VGO dBmrit^ m waveform 
continuously, *e frequency of tfie waveform depending on 


Hfl..3t-11 LFO Modulation df VG0 


VCO 


Pitch 
control 


LFO 


OUT 


Fig. 3.12 Frequency Modulation WaVBftmni 

(a) LFO Waveform: 
(Modulating wave) 


@ VCO Sine Wave 
without modulation 
(carrier) 


Modulaud VCjO wamfemis 



© Shallow; 
(Vibrato) 


® Medium: 
(Exaggerated 
vibrato) 


(e) Deep: 
(Special 
effects) 



Pitch 37 


the sum of the control inputs. The next chapter shovw how 
to start and stop the sound in order to produce separate 
musical notes. 


FiB4 3-13 Prequmey Modulation 


FM RADIO 


RF Carrier 

In m 


RF 

Oscillator 


Modulator 


RF ■ Radio frequency 


Audio 
frequency' 


Audio 
inputs ' 


Audio 
Amp 


X 


Antenna 


Modui«t«ef'M»iMW 




Transminer 



38 Pitch 


3-15 Questions 

1 . What is the diffei^ice l»tween the scale of just intonation and the equally tempered scale? 
2i What Is a beatftequency? Describe how an instrument is tuned using beat frequencies. 

3. What is ^e natural harmonic series? What are tiie musical pitehes of the first harmonics for middle C? 

4. Explain what cau^ consonance or dissonance in a musical interval. 

5. What is an exponential pq^osPBSSidn? 

6. Explain the vdltage-to-frequency/g^|(^ rdation in a linear VCO. 

7. Explain the vbltage-to-frequency/pHsh relation in an exponential VCO, 

8. What is a low frequency oscillator? 

9. What is frequency modulation? 
10. What is vibrato? 


Words to define: 

beat frequency 
cent 

even temperament 
exponential 
exponential generator 
exponential progression 
exponential VCO 
FM 

frequency modulation 

just intonation 

LFO 

linear 

linear VCO 

low fi^quemsy otdlla^or 

modulation 

natural harmonic series 
VCO 

vibrato 

voltage controlled oscillator 


Loudness 


Chapter Four 


■1 Introduction 

The first thing that comes to mind concerning loudness 
is perhaps dynamics (changes in loudness) in music, so 
loudness might seem less important than the other two quali- 
ties of sound. The envelope of a sound might also seem to be 
of not too great importance when trying to indentify a sound 
source, but this is not so. The effect of the envelope on a 
sound is so great that in approaching the synthesis of a sound, 
it is usually far better to synthesize the envelope Isefore 
ccmslderins tone color. 

Before considering loudness as a quality of sound, however, 
we should first discuss how loudness is measured, 

2 MSaiurihg Loudneu 

One way to measure loudness is to measure the pressure 
changes which take place in the air as a result of sound waves. 
Atmospheric pressure is usually measured in units called the 
bar. One bar equals normal atmospheric pressure at sea 
level; or about 70 kilograms per square centimeter (14.7 
pounds per square inch). Since air pressure changes caused by 
even the loudest sounds are too small to be measured with 
a unit as large as the bar, we use the microbar (often written 
Mbar; 1 ijbw = 0.000001 bar). 

Another unit used to measure sound level (among other 

things) is the decibel (abbreviated dB), named after Alexander 
Graham Bell (1847-1922). Most of the confusion associated 
the dedbel Is #ie fesuk of the fact that it is not in 
itself a measure of level and therefore, by itself, is meaningless. 
(Another point of confusion arises from the fact that the 
decibel scale is exponential, rm linear). The decibel is a 
comparison between, or a ratio of two quantities, in this case, 
loudness. If we hear two sounds of different intensities, 
it is possible t& mM6{m thm Bn^ smi that the first Is 6dB 
(for example) louder than the second. At this point we do 
not know how loud either sound is, just that one is a specific 
amount louder than the o^er. 

To me ^e decibel as a unit of sound level measure, we must 

first establish some reference level for comparison; then we 
can say that a given sound is a certain amount above or 
below this reference. 

For acoustic research, aft aneohole chamber is used. This is a 

specially built room with all the walls heavily insulated to 
keep out as much external sound as possible. The interior 
is designed so that all sounds made within the room are 
absorbed as much as possible to eliminate reverberation. 
Besides acoustic research, these rooms are often used by 
manufacturers of microphones and speakers to test amf rate 
their products. 

Using an anechoic chamber, it has been determined that for 
the average person the threshold of hearing, or the point at 
which sound just becomes perceptible, is 0.0002 i^bar. The 
opposite end of pur hearing range is the threshold of pain, 
or the point where sound starts to be perceived as feeling 
(pain). The tiireshold of pain was found to be about 1,000 
/xbars. 


40 Loudness 


Sound pressure level (abbreviated SPL} is a decibel scale which 
uses the 0.0002 ixbar threshold of hearing level as a zero 
reference point The tiire^old of hearing, then, is 048^ I^L 

Fig. 4-1 shows the sound pressure levels of somft ieCKtmnon 
sounds. Note particularly the relation betweftn ■flSe SPL 
scale and the ;ibar scale, and compare this to "tfie relation 
between pitch and frequfeney (Fig^ 3^7). If we insert a test 
signal into a hi-fi amplifier which is capable of producing 
very loud sound, and if we start with the volume control at 
"0" and turn It up at a steady rate of speed in steps of about 
3dB each, to our ears the loudness of the sound will seem to 
increase in steady even steps, much in^e same way as pitch 
increases in a chromatic scale. And like frequMiey, the air 
pressure changes in jubarS will incTvase exponentially. 


Fig. 4-1 Relative Loudness Levels of Common Sounds. 


m = meter I 


Average factory- 
Average automobile — 
Convenation (1ml — i 
Average office — | 

Quiet residential street- 
Average home- 


VBry toftmuslc 1 

Quiet whi sper [2m) 

An echoic chamber 

I l l ll llll — I t 1 1 I I I 


44- 


I I HIM 


Bus interior 
1 — Heavy street traffic (2m) 

I — Heavy truck (35m) 


factory 

■Hammering on steel plate (.75m) 
I — Pneurnatic chipper (2m| 

Threshold of pain 

50-hp siren (40m) 


r 


-HW4 


0.0002 


0.002 


0.02 


0.2 


20 


200 


nnnn Sound pressure 


0 10 20 30 40 50 60 70 80 90 1 00 110 1 20 130 1 40 DECIBE LS 

llll|fi n jlfl MHI l[l M I| M tlj n il M llll n ilh MH l u t MHM [lll.llt M l|llll|II N j LlM I M 


I 1 


fTftt 

—Be 


ickground music 


I — Mmimum street noise 
I I 

' — Very quiet radio at home 


' Country houis 

-Quiet auditorium 
' — Quiet sound studio 

Leaves rustling 
— Thrfishold of hearing 


Loud mutic (rock] 
Thunder 
I — Passing subway 

' Riveting machine (15m) or vary loud music (claislcal) 

' — 10-hp outboard motor (20m) 
lJ6udnfuir6{cbisfeal) 
' — Heavy traffic {10-20m) 
' — Department store or noisy office 


The smallest perceptible change in sound level which the 
average person can detect is roughly 3dB but this will 
depend on the overall level and the frequencies involved. 
In other words. If we produce two separate sounds of the 
same pitch one after the other, they would hwe to ^ 
proximately 3dB different In intensity before iS)e levels 
would seem different. 


4-3 Frsquency ResponsA of the Ear 

The ear is more sensitive to some frequencies than others. 
This sensitivity is also strongly dependent on the loudness 
of the sound. This is shown in Fig. 4-2. The llift margin 0f 
the graph shows sound intensity in dB SPL. Intensity refers 
to the level of sound as measured with instruments. The 
curves on ihe gr^apAi show kiucbietB, whidi i^Xers to the 
subjecti^ level of sound as perceived by tfte ear. Both 


loudness and intensity are measured in dmmt but to 

c^^de at all l«quencfes. the term phon is often used to 
measure loudness. 

Tl °' '^^^^^^ ^''^ ^ short firt spot 

at the ],000 Hz point. This is why 1,000 Hz is used as a 
reference point in these curves (and why recording studios 
often use 1 ,000 Hz as a test frequency or a level reference in 
equipment). In other words, for a frequency of 1,000 Hz the 
oudness level in phons Is exactly the same as the intensity 
eve! m dB SPL. These curves can be taught Of « shovrtng 
the intensity levels which would be necessary to give all 
frequencies the same apparent loudness as the 1 QOO Hz 
reference frequency. For example, to make a 100 Hz pitch 
sound as if it has the same intensity as a 50dB SPL 1 000 Hz 
pitch would mt>W9 an Intensity level of about 68dB SPL 
Ithe^BO phon curve crosses 100 Ht line at about .6808: 

The bottom-most curve on the graph, the 0 phon loudness 
curve, represents the threshold of hearing for various fre- 
quencies. At 1 ,000 Hz. the threshold of hearing is OdB SPL 
Lower frequencies, however, require higher intensities before' 
they start to b» hearti: l^of example. 100 Hz requires a 
level of 38dB SPL before it starts to be heard, while a fre- 
quency of 30 Hz requires an intensity of more than 60dB 
bKL. From this itcan be seen why soft bass passages will cause 
a tape recorder VU meter to jump higher than treble passages 
of the same apparent loudness. 

These curves, of course, represent average values. They would 
ook slightly different fof diffemnt people and slightly dif- 
ferent for the left and right ears. This would perhaps explain 
why people like different settings of amplifier tone controls 
One might wonder at a friend who likes to turn the treble 
o^rol up too high; it could be because his/her hearing is 

Another point in connection with these curves is that most 
knobs which control level in electronic equipment control 
the mtensity rather than the loudness. Thus, it is quite pos- 
sible to set a control low enough that the lower frequencies 
are below the threshold 6f heaHng. Some hi-fi amplifiers 
have a "loudness" control which theoretically can be set at 
any point and still retain the same apparent loudness for all 
frequencies So that with low listening levels they are not 

Since the loudness of different frequencies varies with the 
setting of the system level controls, it can be seen that it 
would be extremely importfintduringthe process of recorifing 
electronic music to monitor the output at the same level at 
which the finished recording is expected to be played. 

Dynamic Range 

The dynamic range of orchestral music can exceed lOOdB In 
0^ wo^ if we call the level of the softest passage 
uau. the loudest passages can ex(»ed +100dB. Hectronic 


42 Loudness 


recording and reproduction equipment can seldom match 
this performance. The upper limit in such equipment Is 
dtftefmined by the level of signal which can be processed 

before distortion occurs. The lower limit is determined by 
the noise level of the equipment. All electronic equipment 
generates a certain amount of noise. This noise can be heard 

as a hissing sound when the system volume controls are turned 
up high when the system is not processing an audio signal. 

The usable dynamic rangie of an ordinary good quality open 
reel tape recorder is on tfte order of about 5& to 60dB. 

This, of course, Is it0iii^iQYe;;(i6ar the lOOdB dynamic ran^ of 
live music. A little analysis will show that this is not; bad 
as it seems. If we assume an average home with a backgfdund 
noise level of about 45dB SPL {Fig. 4-1), it would be neces- 
sary for the music level to be either the same or more than 
this for It to be heard. Any lower level would be masked or 
covered by the background noise. If we add a lOOdB dynamic 
range to this 45dB SPL background noise, we get a top level 
of 145dB SPL for the loudest passages in the musfc. Ignoring 
the effect this would have on neighbors, this level is well 
above the threshold of painl Assuming a conservative dynamic 
range of dfobut SOdSfbr a fape r«cor3frig, the liDudest passages 
would reach a level of 95dB SPL, a little more realistic level 
if the listener likes very loud music and has understanding 

Fig. 4*3 shows two audio waveforms. In (a), the input level 

of the signal has been adjusted so that the peaks are just 
inside the maximum permissable level of the equipment 
processing the signdi. Tlw dotted line shows the average level 
which represents its apparent loudness. In (b), even with the 
peaks going over the maximum permissable level (thus 
distorting tha sl£piai)r the sv«r^ level and the apparent 
loudness is lower than that of (a). 


Fig. 4-3 Relative Sound Levels 




Maximum 

permissible 

level 

Approximate 



® will sound louder than (£) even though thepMlcs titf ® 
are much higher then those of (|). 


4-5 Intensity of Harmonics 

The tone color of a sound' source Is determined by the 

number of harmonics contained in the sound and their 
relative intensities. This is another aspect of loudness as a 
quality of sound. The importance of the Intensity of the 
harmonics in a sound can be demonstrated by examining the 
harmonic content of two waveform* opmmonly used in 
sy nthesii : ^ square warn and the ^tangle vMve. The speetnim 
diagram for the square wave is repeated in Fig. 4-4 for easy 
comparison with the spffmim diagram of the triangle wave. 
Note that, ffitccept for the intensity of the harmonies, these 
waveforms contain the same harmonics: all the odd num- 
bered harmonics. In the case of the triangle wave, the 
intensliy df the «)dst!ng harmonics is so tow tiut the triangle 
wave has a tone quality very near that of the sine wave; a 
tone quality very far from that of the square wave. In Chapter 
i it will he shown how the square vrave can be Mterad so 
ihat it very closely approaches the shape of ^e iblangle wave. 


Loudness 43 


Fig. -M SYNTHESIZERWAVEFORMS 
















© 




















HARMONIC SPECTRUM 
OF PERFECT SQUARE 




















WAVE 






































n 



















-10 
-20 

— .tn 





















































-40 

-50 


















































HARMONIC 




3 


5 

7 


9 


H 


13 

15 

17 

1 

a 

FREQUENCY 

X 


3X 

6X 7X 

9X 

11X 




ISX 

AMPLITUDE 

A 


A/3 

A/7 

A/9 

A/1 1 



"■ A/18 

dB 

0 


-9.5 


14 

■17 


-19 


-21 





Note that the denominators of tiie 9rnp|;tude fractioni above ars the same as the harmonic number. 


HARMON IC SPECTRUM 
OF PERFECT TRIANGULAR 
WAV6 




\ 




/ 

/ 

/ 

\ 

\ 






0 

-10 








/ 








S 

\ 






























-20 







































dB -30 

-40 

-50 
-60 
































































HARMONrC 

1 



3 


1 


7 



n 

13 15 17 

9 


FREQUENCY 

X 



3X 

5X 7X 

9X 

tlx 

13X 15X 17X I9X 

AMPLITUDE 

A 



A/a 

A/2S A/49 

A/81 

A/121 A/iraA/22Sll/a9Aj3It 

dB 

0 



-19 


28 




-42 


45 - 

47 -49 -51 


Note that tiie denonrinttOfS^^ aM^ucifrlrsc^ the s 

qu 


H Of the iian$toiiii& humtiigr . 


44 Loudness 


4^ Envelopes 

The voltage controlled amplifier or VCA is the primary source 
of level control during the production of notes on the 
syntherizer* The term "amplifiiw" might actually be con- 
sidered a misnomer since in many synthesizers the normal 
output level of the VCA never exceeds the input level. The 
important point, however, is that the output leva! ts ^trolled 
by an externally applied control voltage. 

The shaping of the loudness contour or envelope of a sound 
is done by controlling the VCA with an emrelope gen- 
erator (sometimes called an ADSR, named after its 
controls^ or sometimes called a transient generator). The 
output of the envelope generator Is a cmtrd iKittageWhii^. 
when applied to the control input of the VCA, gives the 
sound its loudness contour. 

When a key on a piano is struck and held down, the resulting 
envelope will be (ike that shown in Fig. 4-5 (a). Th^ thtie 
required for the sound to jump up to maximum is called 
attack time. The time required for the sound to die away is 
called decay time. If the piano key is released before the 
sound has had a chance to die away completely, the sound 
will be quickly dampened, as shown in (b). This is called 
release time. 

Fig. 4-6 shows two envelopes possible on modern electronic 
organs. In (a) is shown an envelope possible only on electronic 
instruments. As shown, attack and release tlm^ are 2ero, an 
impossibility. In actual practice, these are probably on the 
order of a few milliseconds or so (1 millisecond = 0.001 
second). The very fast attack and decay cause this envelope 
to produce a very artifical electronic sound; such sounds 
normally do not exist in nature. In (b) is shown an envelope 
with slightly longer attack and rel&as^ times used to fmit&te 
the start and stop of the air flow through an organ pijse. Both 
these envelopes introduce the element of sustain since the 
sound will continue as long as the related key Is held down. 
Notice tiiat in botii (a) and (b), decay is missing. 

Fig, 4-7 shows a synthesizer envelope which contains all 
four elements of the envelope: attack (A), decay (D). sustain 
(S), and reiease (R). Pressing a key on the keyboard produces 
a gate pulse which triggers the beginning of attack time 50 
that the level of the VCA output sound begins to rise. When 
the sdund reaches its-maxfrntmi Idvel ift ^6 end tif attack 
time, decay starts and the sound falls to the level determined 
by the sustain control setting. This level is then maintained 
until the key is releioed. 'Releasing the feeV ojts off ^e gate 
pulse and triggers the beginning of release time where the 
sound finally dies away completely. Note that if the sustain 
cdiml ^«Btstimimum, as wh^ aymhesizing tfte envelopes 
jthown in Fig. 4-6 (v) and (b), the decay element is missing 
and the decay corrQ^ol has no effect. The envelope shown in 
Fig. 4-7 might be used for synthe^^riit brass irmjment 
sounds, or any instrument in which aiji%p peraisu've playing 
style is desired. 

Fig. 4-8 the envelope generator output When synthe- 


FiB. 46 Pianipf irmlllCMft 
(a) Piano envelape 



PtarioehmloiSa fdampariodi 
Key 
struck 



A" Attack time 
D - daisy ^nio 
R " Release time 


Fig. 4-€ Organ Envelopes 

® Electronic 
Key 

ON 


Key 
OFF 


(St Pii»4>rsen 
tCay 
ON 


-R - "0" 

S " Sustain levei 




sizing piano-like envelopes. Note the differences between 
these and the envelopes shown in Fig. 4-5. The curves formed 
by tile synthesizer envelopes are exponential curves. Experi- 
ments have show/n that exponential attack, decay, and 
release curves sound natural and that trying to generate 
small fluctuations encountered In natural sounds ssmns to 
add nothing useful to the sound as perceived by the ear. 

VGA Control Response 

One frnportant corttideration about the VGA Is Its response 

fn relation to the input control voltage. In most VGA's, the 
response is linear; a change of one volt in control voltage will 
cause (Hie volt change tn 0w output signal leveL Kg. 4-0 
shows the control response for a given VGA with linear 
response. For example, if the control input is 2v, the output 
Will also b6 2v. In (bh the output waveform is a faithful 
reproduction of the loudnea contour at the control voltage 
input. 



Fig. 4-8 VCACbmiDl Response 
0 Unear Response 14 

12 


10 


Oitput 
voltage 


(§) Waveforms 




Audio 
OUT 




Exponential 

deesy 

lOv 1 


1 


-A 

A 

-* ft-j 


Some synthesizers give the option of being able to select 
between linear and exponential response. Fig. 4-10 shows the 
output of a given VGA wtth exponential response. Mote 
that in {a), the output level is shown in decibels so the 
graph forms a straight line. The response is exponential 
because dedbel scale Is exponential For comparison, (b) 
shows the voltage outputs for the same VGA. In this the 
exponential response can easily be seen. As an example. If 
the GOntrdt Input is 6v, the output vvlll 'be -40dB ss shown 
in (a),or0.1v, as shown in (b). In (c) is shown the waif^Otms. 
The signal and control inputs are the same as those ^own in 
Pfd- 4-9 ib), but notft ^e marlced difference in the output 
waveform. The control input represents the exponential 
decay of an envelope generator. The output in Fig. 4-9 (b) 
is a faithful copy of this expPnanttiif decay. Due to ^e 
exponential response of the VGA in Fig. 4- 10, the exponential 
decay of the control input produces a very exaggerated decay 
iln itNt^i)Etpit|.s«t J&l^lilils is very useful for symfiestzing 
^arp, percussive sounds. 


Fftf. 4-10 VCA Control Response 


® Exponential Response (dB Output) (g) Exponential Response (Voltage Output) 



Control voltage (V) Control input 


Q Waveforms 



I I apEWda^ws "0" in the ou^t 

A ' 


Loudness 47 


4-8 


Amplitude Modulation 

Using the VGA to control the level of a source signal is a 
-brm of ampittudfr mixiiMoii (AM), the same principal as 


Fig. 4-1 1 Amplitude Modulation 


RF Carrier 


RF 

Oscillator 


RF • Radio frequsney 


Modulator 


Modulated Carrier 
Audio \ 

1' w %w 

\ 


Antenna 



Transmitter 


Audio 
inputs 



Audio 


frequency 

Audio 

Amp 



4-9 


is used In AM radio broadcasting (see Fig. 4-11). In music, a 
regular wavering of sound level is called tremolo. This type of 
amplitude modulation can be demonstrated by using an LFO 
as the control input to a VGA, as shown in Fig. 4-12. Since 
with no control i nput to the VGA, the VGA remains "closed", 
It is necessary to supply a fixed voltage to the modulation 
input so the output level remains at half Its normal 
maximum level. In this way, the VGA will react to both 
upward swings {+) and downward swings {— ) of the LFO 
wnveform. The upward swings will cause the VGA to "open" 
more and the downward swings will cause the VGA to "close" 
more. Most VGA's provide what Is called an initial gain 
control (or sometimes hold control) which can be used to 
hold the VGA in a partially "open" condition. 

Fig. 4-13 shows the waveforms' produced by the above 
an^ngement. In (b) Is shown the VGA output with no LFO 
modulation but with a fixed voltage (or initial gain control) 
holding the VGA partially "open". In (c), (d), and (e) can 
be seen how varfous depths of LFO modulation cause the 
VGA output level to periodically waver above and below tfie 
level set by the fixed voltage input. 

The Basic Synthesizer Patch 

Fig. 4-14 shows the basic synthesizer patch with the addition 
of VGA modulation by means of the envelope generator. 

A voltage appears at the keyboard gate output anytime a 
key fs deptmed. Since this Voltage goes m and off with the 
pressing of keys, it is called a gate pulse. The gate pulse is 
most often used to trigger the envelope generator into opera- 
tion. The ou^t of ^e envetope genen^r #ten "opens" 
^e VGA to let sound out of the synthesi^r. 

The next chapter deals with ihe control of tone color in the 

output sound. 


Fig. 4-12 LFO Modulation of VCA 
I 1 

— H VCF h 


VGO 


L I 

Fixed 
voltage 
source - 
(See text) 



OUT 


LFO 


Fig. 4-13 Amplitude Modulation Waveforms 
(a) LFO waveform 

0 


® VCA output 
without LFO 
modulation (carrier) 

Modulated wavefomn: 

(£) Shallow: 
(Tremolo) 


@ Medium: 
(Exaggerated 
tremolo) 


(e) Deep: 
(Special 
affects) 



48 Loudness 


Fig. 4^14 The Basic Synthesizer Patch 


( — ^-1 

I LFO I ' 

1 I 
I I 


Key 
CV 


vco 


VCF 

mm 






Envelope ^**~\^ 


Kevboard 
gate 

pulse 

_rL 





II!! 

Ill |i 

i| III! 

Ill 


4-10 Questions 

1. How is loudness measunsd? 

2. When recording eiectrontc music, why is it importffit to monitor the music at the samfe level at which the 

finished recording is expected to be played? 

3. What limits the dynamic range possible in a given piece of electronic equipment such as a tape recorder? 

4. What harmonics are contained in the square wave? In the triangle wave? What is the difference in harmonic 
content between the two? 

5. Draw the envelope produced by striking and releasing a piano key. Draw the envelope produced by a pipe 
organ nou. Name the parts. 

8. }f the sustain control of an envelope generator is set at maximum, what effect will the decay ccmtrol have? 
Why? 

7. Show the output waveform of a linear VGA wilh a sine wave input controlled by a piano-tike envelope. 
Show the output of an exponential VGA under the same conditions. 

8. What is amplitude modulation? 

9. What is tremolo? 

1 0. Draw a diagram of the basic synthesizer patch and explain how it worlcs. 


Words to define: 

ADSR 

AM 

amplitude modulation 
attack time tA> 

dB 

dB SPL 
decay time (D) 

decibel 
envelope 

exponential decay 

exponential VGA 
gate pulse 

intensity (of sotHid) 

linear VGA 
loudness 
mierobaf ifi\>6t\ 

phon 

release time (R) 
sound pressure level 

SPL 

sustain (S) 
^reshold eif hearing 

tremolo 
triangle wave 
VGA 

voltage controlled amplifier 


Introduction 

Timbre or tone color is that quality of sound which allows 
us td distinguish between two sound sources producing 
sustained sounds at the same pitch. Tone color is usually 
very much affected by the pitch range and the loudness 
contour (envelope) of the sound* so It is u»»lty better to 
consider tone color last when synthesldrtg sounds. 

Noise 

All electrdnie eltmiHS deiiei^tteittitifi emounf c»f noUe^ 

in most cases tflfe notse is undesirable. In electronic music, 
however, noise Ofl^n forms the starting point for tone color 
in the syntiiMlis of ntm^pIMied sounds or fbr n^^t^ed 

elements In sounds. In electronic music we are concenred 
with two types of noise: white noise and pink noise. 

Like white light, which is a combination of equal amounts 
of all colors, white ncrfse fs a cdmbination of equal amounts 

of all audio frequencies and produces a hissing sound. A 
review of Fig. 3-7 will show that the number of separate 
frequencies contained in eai^ octave Is doi^e the number 
contained in the next lower octave. For example, the octave 
between Aj (110 Hz) and A3 (220 Hz) contains 1 10 separate 
frequencies (excluding fractions) the next higher oetatve^As 
(220 Hz) to A4 (440 Hz), contains twice as many frequencies: 
220. Each successive octave, then, contains twice the number 
of frequencies as ^fr ne9Ct tower octave. A doubling 'crf 
frequencies for each octave means a power increase of 
3 decibels per octave. Since there are more of the higher 
frequencies, these frequencies dominate the sound to give it 
its characteristic hissing sound. This is aided by the fact that 
the ear is less sensitive to lower frequencies. 

Pink noise is noise which contains equal amounts of energy 
in each octave. Since each octave (rather than each frequenoy) 
is equal in energy, this type of noise sounds, to our ears, as 
if it has an equal amount of all frequencies and produces a 
rushing sound much Mice a waterfall. 


Filters 

The major source of tone color in the voltage controlled 
synthesizer is the VCO for pitched sounds and the noise 
generator for non-pitched sounds. The major source of 
control over tone color changes are the filters, the most 
common of which is the voltage controlled filter or VCF. 

In Chapter One It was shown how a pipe or tube is resonant 
at various frequencies. The fact that this is so makes the tube 
a form of acoustical filter. If we speak into the end of a tube, 
the quality of the voice emerging from the other end will be 
quite different. The human voice contains a great many 
different frequencies. When one or more of these frequencies 
coincide with any of the resonances of the tube, these 
frequencies are accented in the output sound in exactly the 
same way as the sound of a tuning fork will be accented 
when its frequency matches any of the tube resonances. 


The tube, acting as an acoustical filter, alters the harmonic 
content of any sound which passes through it. Changing the 
length of the tube cKartgw the r«onanoe pcM'nts and, there- 
fore, the tone color of the sound emerging from the opposite 
end. Rooms also act as acoustical filters since they affect any 
sound made within them. 

tn this text, we shall be primarily ^jneeiTii^ with four tHBic 

types of filters commonly used in electronic music: the low 
pass filter (LPF), the high pass filter (HPF), the band pass 
filter (BPF), and the band reject filter. The names imply the 
functions. The low pass filter passes low frequencies and 
blocks high frequencies, the high pass filter passes high fre- 
queneiesv and ^e band pass fitter passes a band or group 
of frequencies while the band reject filter rejects a band or 
group of frequencies. The most common form of VCF is 
the voltage controlled low pass filter. This type of VCF is 
so common that when we hear or see the term VCF, it is 
usually safe to assume that it is a low pass filter. Many synthe- 
sizers also provide a high pass filter; in some systems, this 
high pass filter is voltage controlled. Larger systems also 
sometimes provide for voltage controlled band pass filtering. 

54 The Low Pass Filter 

Fig. 5-1 shows the frequency response of a given low pass 
filter (LPF). The OdB level is a reference which represents the 
nomial output level of the filter when a signal of a given 
level is applied at the input. The actual levels involved will 
depend on the design of the particular filter. The shading 
shows the range and levels of frequencies which can be 
passed by the filter. For example, if we apply a 100 Hz 
sine wave of the correct level to the Input of the filter, 
the output will be OdB, or normal. If we change this sine wave 
to 1 ,000 Hz without changing the input level, the output will 
be -20dB, or 20dB below the OdB reference level. Again, 
changing the sine wave to 11,000 Hz, the output level will 
be below — 60dB. This level Is so low in relation to the OdB 


Fig. 5-1 LOW Filter 

Frequency response Cutof 

f point 

-10 

dB -'0 

-30 
-•0 

























/ Cutoff 
/ point 











































































-3dB ] 









Hi 























1 100 

mnv 


























HI 























Sini 



















































































i 


w an 3D 

.0 70 ^oa 200 300 no m IAD MOD uoo ftDoo nooo nues 
Frequency 



reference level that we may consider the 11^00 Hz sine wave 
as being missing at the ou^^jt fttter. tn many fitters, 
this would probably be below #8 nolle floor of the filter 

anyway. 

For lower frequencies, the response of the filter is flat; no 
matter what the frequency of the sine wave input, the output 
will be OdB {assuming the input level remains as before}. 
Remember that it requires approximately 3dB difference in 
level befbre ^e ear can detect a dffferenee in loudness; 
therefore, a sine wave of about 320 Hz, which in Fig, 5-1 
produces an output of — 3dB, can be considered the upper 
limit of the flat portion of the response. This point on the 
filter's curve is called the cutoff point or cutoff frequency; 
any frequencies above this point will produce an output level 
below -3dB, or levels not as loud a»loM^ fnvtiMi^es. W&h 
filters used in electronic rmitte, ^if euttfff ^ almost 
always adjustable. 

The rate of fall-off of the filter slope js an important con- 
sideration because i\ affects ^e relatTve tevels of the fj^ 
quencies above the cutoff point. The slope is measured by 
determining the amount of level change which occurs for 
one octave change firfrequency. thfs^measurement is taken 
from a portion of the slope which is representative of the 
whole. In other words, measuring the slope at the cutoff 
point WQiib^ ^ tnaeeurtite slhoe the slope Is curved at this 
point, fsbom I^MMJt 500 Hz in Fig. 5-1, the slope falls in a 
relatively s|3^(^l line. For a 1,000 Hz sine wave input, the 
output level Is ~20dB. A sine wtm one octave above this, or 
2,000 Hz, would produce an outptlS of -32dB, an output 
level which is 12dB less than the^ivtsine wave. The slope of 
filter, then, is -12dB/8va (—11^ decibels per octave). 
The minus sign indicates that the slope falls for an increase 
in frequency. A plus sign would indicate that the slope rises 
for an IncreKe in frequency. The most common filters, at 
least as far as electronic music is concerned, are — 3dB, 
— 6dB, — 12dB, and — 24dB per octave. Pink noise is usually 
generated bY passing vfhite noise through a — 3dB/8va low 
pass filter, as shown in Fig. 5-2. -12dB/8va and — 24dB^va 
are the most common slopes found in voltage controlled 
fitters. 

Fig. 5-3 shows the frequency response of what could be 

termed the perfect low pass filter because alt frequencies 
above the cutoff point are missing from the sound and all 
fraquend^ below the cutoff potnt are m the OdB reference 
level. In the case of electronic music this type of filter is 

probably undesirable. 

Up to this point we have been primarily concerned only with 
ihe effects the low pass filter on sine waves of given fre- 
quencies. In subtractive synthesis we are usually more 
interested in the effects of filtering on waveforms rich in 
harmmilcs. Fig. 5-4 (a) gives a repetition of the spec^^ 
diagram for the sawtooth wave. Let us assume a sawtooth 
wave with a pitch of Aj (bottom space bass cleft) passed 
tiirough a low pass filter with a cutoff point set at about 300 
Hz. The results can be shown by superimposing thespectrum 
diagram of the sawtooth wave onto the frequency response 


Flg.B-2 Nolte Generator 


Noise 
Source 


-3dB/8va 
Low 
Pan 
Filter 


■White nt^Mout 


•Pink noin out 


Fig. ''nrtaet" MMv M Ftltvr 

Cutoff point 


0 


dB 


-40 ' 1 ' I I I M U 

FraqMeney 


Fig. S-4 Effe^ 0f FIKefing Hatimmics 

@ Harmonic contmttif sawtooth wave 

0 

-10 

-20 

dB -30 

-40 

-SO 

-60 I 

HARMONIC NUMBER 1 2 3 4 5 6 7 8 9 10 — N 

FREQUENCY X 2X St 4X 5X 6X 7X ax gx lOX 

© Hamwnicsof 110 Hz sawtooth wave luperimpotsd on low pan filter frequancy response curve 


dB 


D 
-1D 
-30 
-30 
-40 















-l-n 

Cl 

i-r 
t 

-1 

3ff poin 

1 

^3 

1 1 

00 

1 — 1 
H 

E. 







































































































\ 





























1 

f 
1 

■ ' 



























1 































1 1 1 J l\ 




























1 II , 1 ' 1 





















HARMONIC 







1 

1 1 1 1 1 





















NUMBER 





il 
il 

1 


( 
























1 

■ 


k 














* 

h 


«' 




BU'' 

« 

0« 


M 

r 






FREQUENCY 


curve of the filter, as shown in Fig. 5-4 (b). For comparison, 
the dotted lines show the levels the harmonics would reach 
without filtering. As can be seen, the first harmonic is not 
affected, but all ihe other harmonics are attenuated to one 
extent or another. It will be seen in (a) that the normal level 
for the fourth harmonic in the sawtooth wave is — 12dB in 
relation to the fundamental. According to the filter's response 
curve as shown, the output level for a 440 Hz sine wave (the 
frequeii^ fourth harmonic) would be — 6dB, in 

o'tfier wbrds it would be attenuated by 6dB. The fourth 
harmonic, then, is attenuated by 6dB so that its level is— 18dB. 
The levels of the other harmonics are affected in a similar 
"feshion. Since the levels of the harmonics in the filtered 
savirtooth wave are different from the original levels, the tone 

color and the shape of the waveform will also be different. 

..... 'v ■ : 

(Effoets of fiiisring' on waveforms is discus»d in more mml 
later in 



The Voltage Controlled Low Pan Fitter 

It can be seen in Fig. 5-4 (b) that as long as the cutoff point 
of the filter remains fixed, the harmonic content of the out- 
put sound will remain fixed only if the pitch cif #i#sawtd&^ 
wave also remains fixed. Changing the pitch of the sawtooth 
wave would effectively move the spectrum diagram to the 
right or left in relation to the filter response curve, dianging 
the harmonic content of the output sound. In other words, 
playing a chromatic scale will produce notes each of which 
has a different tone color. If we play the pitch of 880 Hz 
As (one ledger line above the treble cleft), the level of the 
fundamental will have fallen off more than 15dB. Going 
mud) higher will considerably reduce the output sound level. 

Like the VCO TUNING control, the VCF CUTOFF FRE- 
QUENCY control sets the initial cutoff point of the filter, 
after which ft can be altered with externally applied control 
voltages. If we use the keyboard control voltage output to 
control the cutoff point of the VCF. It is possible to cause 
the filter cutoff point to follow the pitch. In the example 
given in 5-4 above, the filter cutoff point was set at the 
approximate frequency of the third harmonic of the 1 10 Hz 
As sdWtOO^ WdVB. By applying the pitch control voltage to 
the VCF, it is possible to cause the cutoff point to move 
with the pitch and remajj^ at the approximate point which 
represents the third harmohiccyTany pitch played. From this, 
logic would seem to dictate that it would be essential that 
the VCF have exactly the same voltage-to-frequency relation 
as the VCO. If a change of 1 volt causes one octave change 
in VCO pitch, it should also cause one octave change in 
the VCF cutoff frequency. With this relation, the VCF can 
foltow d^i^ly any flanges in pitch in the music being 
played. 

The tone color of most instruments changes wi'tti pitch. 
Usually the higher pitches are brighter than the lower pitches. 
Since high harmonics of high level produce a bright tone 
color, this might seem to imply that if a chromatic scale is 
played up the keyboard, instead of exactly following the 
pftch of the music, the VCF cutoff point should actually 
anticipate the pitch by moving higher than normal so as to 
allow more harmonics to pass. Instead of 1v/8va it might 
seem that something on the order of two octave change 
of the cutoff point should occur for 1 volt change in pitch 
control voltage. In actual practice, because of the peculiarities 
in our heartng, this is not tt6cimt9. In eS(perimenf(ng wf^ 
1v/8va filters, it will be found that if the pitch control volt^BS 
is attenuated by approximately half, the tone color of the 
output sound wilf seem to our ears to retain an approxfmately 
constant harmonic c(ȴ|efit if the pitch control voltage in 
not attenuated, the tone color will seem to get brighter with 
higher pitches. A response Of iMra fiMitits vfitid. 

The tone eotdf 0f many instruments, especially wind instru- 
ments and plucked strings, changes during the production of 
each note. This can be imitated by using an envelope genera^ 
tor to cause the VCF cutoff pointto change. ©fHm^itaf 

envelope which is used to control the VGA can also beim^' 
to control the VCF. In other cases, it is advantageous to u^ a 


56 Timbre 


second envelope generator so that the sweep of harmonics 
is independent of the loudness contour. Some synthesizers 
allow tm^i^ion 6f tftg dnvtlope so tiiat the harmonics can 
sweep in the opposite direction from what is nortrtalty 
expected. 

Another common source of modulation fiar the VCF h ite 
LFO. whidi produces a wavering of the tone color c^ed 
growl. 


5-6 The High Pass Fitter 

The high pass filter (HPF) may be thought of as an «Xaet, 
mirror image of the low pass filter. Fig. 5-5 (a) shows '^e 
frequency response of a high pm filter viAth a $I0|» of 12dB/ 
octave. In (b) is shown the spectrum dia{NWn of 8 sawtooth 
wave superimposed on the response curve of ^e filter. In this 
flxamplft, ^e frequency ^« sawtooth wave is 1 10 Hz and 
the cutoff point of the filter is set at about 770 Hz which is 
the frequency of the seventh harmonic of the sawtooth wave. 
The smmitfi haitnorilc is Hhia sirorigsit, while ftmdwnantal 
is attenuated by almost 35dB. The third harmonic is normal- 
ly 9.5dB lower than the fundamental. It is attenuated by 
16dB due to the filter's slope so that its level becomes 
-24.6dB. 


Fig. 5-B HlghPaw BHBF 
(§) Frequency response 


Cutoff point 


























} 












t ■ 




















■ 1- 










































































































































n JOB Mt no Mw - 
FREdUENCV 


dB 



aoo aw no na vm' 
FREQUENCY 


Experiments have shown that if the ear is simultaneously 
given a series of overtones which fall within the natural 
hannonic series, it wi!t h^r ^e fundahfidrital pitelh even ff 
the fundamental is completely missing from the series. This 
explains the 16 Hz organ pitch shown in Fig. 1-2 which falls 
below *e normal f»«c(ueney range the ear. The ear hears 
all the harmonics which are within its normal range and 
"mentally" furnishes the fundamental which is too low to 
hear. This phehomehon alto ex^^tw why a sdU'nd with averi 
a great amount of high pass filtering usually does not lose 
its previous pitch sense. 

Since high pass filters block lower frequencies, they are often 
used to brlghtsn sy»lhestz«d soun^. Soma synthesizers 
include voltage eontrdlled high pass filters. 


The Band Pass Filter and Band Reject Filter 

The band pass filter (BPF) passes a band or group of frequen- 
cies. The average band pats ffltarhasa naritnv band wld^ as 
shown by the response curve in Fig. 5-6 (a), the only dif- 
ferences being in the slopes. In this example, the center fre- 
quency of the filter Is 1^0 Hz so that fi^uenelM higher 
and lower than this will be attenuated. The response for a 
wide band pass filter is shown in (b) (next page). In this filter, 
the response between the upper and low«r l^tts tt flat A 
wide band pass filter of this type can be made by (IMII^Inga 
low pass filter in series with a high pass filter as Aown Ih (c). 
In this case, the control of the upper and lower Hfflfts m 
independent of each other so that the limits may be set at 
any desired point. 


Fig. SS Band Pan Filters 

(a) Narrow bandwidth 



FREQUENCY 


SB TiiAbre 


(Fig. 5-6) 

(g) Wide bandwidth 


dB -» 











































1 























\ 













3dB 





















—3 dB 














1 

1- 























4— 






















B 


a 


i( 

1 

































ir 


fl 


















































































































































































































































































FReaUENCY 


© Makiirg* band pass IIHsr 
LPF R«n>Mue 


HPF Response 


BPF Response 


IN 



Low 

Pass 

Filter 


High 




Pass 
Filter 



GUT 


The result will be 
the same if the 
two filters are 
reversed 


The band reject filter is an inversion of the band pass filter, 
as ^own in Fig. and can be made by patching a low pass 
filter in parallel with a high pass fil^i: as shown in (c). 


Effectt of filtering on Waveforms 

It is of a certain value to demonstrate the effect that filtering 

lias on a given waveform. For this, the square wave is ideal 
twcause it contains the extremes of both high and low fre- 
quency (»mponents. High frequencies are characterized by 
rapid changes in voltage level. In the "perfect" square wave, 
the vertical edges would be instant changes in voltage, an 
impossibility since they would then represent a frequency of 
infinity. Even a medium quality square wave, however, will 
produce vertical edges which would be equivalent to very 
high frequencies. Low frequencies are characterized by slow 
(from the audio point of view) changes in voltage. The 
horizontal portions of tiie "perfect" square wave would 
represent zero Hertz (DC voltage, or no dtjanges). 

Fig. 5-8 (b) shows what happens to a square wave when it is 
passed through a low pa^ filter. The tow pasa filter can be 


Timbre 59 


Fig. 5-7 Band Rejeet i^tesnt 
0 Ntlrtowbaindwidflfi 


dB 


































\ 







/ 























} dB 

< 











































































































B 

an 

dwidth— 
























































- 



































































'4i 4) N M M> m Me 100 TOO iju vm xooo 
'FREQUENCY 


MOO TMS lOAM 


(b) Wide bandwidth 







\ 








E 

MM 





























T 

S 
















7 















-3cl 

8 












































































































































































































































dB 


10 in 30 u ro 100 


300 300 goo no ijxn imo voa »mo i.mo io^ 

FREQUENCY 


© Making a band reject filter 

LPF Response + 


IN- 


HPF Response 


Band Reject response 



Low 
Pass 
Filter 









1 



High 
Pass 
HIter 


1 





OUTPUT 


60 Timbre 


5-9 


thought of as being slow and sluggish, resisting fast changes. 
This tendancy causes the output of the filter to lag behind 
tfie quick changes in the square wave, softening the comers 
of the squares. The lower the cutoff frequency of the filter 
in relation to the frequency of the square wave, the more 
pronounced the softening of the high frequency component 
vertical edges will be. Notice that with the cutoff point low 
in relation to the input frequency, the square wave begins to 
toolc very much like a triangle or sine wave, indicating the 
supression of the upper harmonics. 

Fig. 5-8 (c) shows what happens to a square wave when it is 
|^$ed through a high pass filter. The high pass filter can 
react very quiclcly to sudden changes in a wavefomi, but 
it can be thought of as being very poor at holding fixed levels 
as occur in the horizontal portions of the square wave. 
Since these horizontal portions cannot be hieild. they tend 
to fall back towards the zero level; the higher the cutoff 
point of the filter, the faster these horizontal levels return to 
the zero tevel. With Ifie cutoff point viary high, the filter pro- 
duces a waveform with only sharp spikes since it effectively 
passes only the high frequency vertical portion of the square 
wave and blodcs all of tfta hcMrteontat touf':;fi9fipieney portions. 

The eontrol voltage output of the keyboard controller Is 

very much like a square wave In that when a series of pitches 
is played, the output voltage makes very quick jumps up or 
down to each new level. Fig. 5-9 (a) shows the trontrol voltage 
output for a chromatic scale. In (b) is shown what happens 
to this control voltage when we add the effect of portamento. 
'Ihis demonstratas the faet tl^at the portamento dibits are 

basically nothing more than a tow pass filter applied to the 
keyboard control voltage output. If we play a trill on the 
keyboard, the control voltage output will be a tow frequency 
square wave. Using various depths of portamento will produce 
waveforms exactly as shown in Fig. 5-8 (b), the only dif- 
ference Isaing that the related frequencies would be much 
lower. The frequency produced by even a very rapid trill 
on the keyboard is very low compared to ordinary audio 
frequencies. Most filters, except of course thosa desifhed to 
produce portamento effects, have a low frequency limit 
which would be far above these keyboard "frequencies" so 
that ev«i a low pass^ter would act as a high pass filter on 
the keyboard eormidlVdiltage output 

Resonance in Filters 

In aleetfoniM, feedback refers to fhe frading of a portion of 

the output signal of a device back into the Input of the same 
device. The result will depend on whether the feedback is 
pdsltiva or n<^aUva. 

In negative feeicttiaak, tha polafify dr phase of '^a-feedbaek 

signal Is opposite the polarity or phase of the input signal; 
the result is a dampening of the input signal. Many audio 
amplifiers use negative feedback intemalty. Tlie n^f^^ 
feedback tends to cancel out distortion and noise which occur 
within the amplifier, thus improving the quality of reproduc- 
tion. 


Fig. 5^ Eftoeti of Filttring on WaviEifdrms 
@ Square wava 



@ Low pass filtering 


CO 

Freq 

high 


CO 
Freq 
medium 
high 


CO 
Freq 
mecBum 
low 

GO 

Pi«q| 

Low 



©High pais filtering 


CO 
Freq 

tow 


CO 
Freq 
medium 
low 


CO 

Freq 

mKUym 



CO 

Fraq 

high 


Timbre 61 


Fig. 5-9 Keyboard Portamento 

Keyboard control voltage ou^t for chrometic scale 

Voltage 
6/12 


@ Portamento 


5/12 
4/12 
3/12 
2/12 
1/12 


1/12 i— ' 



Time 


In potittve fMidbMik, ftddbadc signal is of thd same 

polarity or phase as the input signal. The feedboelc signal 
reinforces the input signal. It is easily possible fer this 
positive feedback signal to reirjfbrce the Input so much ijnttt 
self-oscillation occurs. The classic example of ^fs Is the 
howling PA system in which sound from the Sjiraksr re* 
enters the microphone to b« aimpHfldd^ dv6r arid t/m ogain. 
Many types of electronic oscillators use pdSltEve fetfdbttok to 
produce oscillations. 

Resonance in filters is produced by inducing positive feed- 
'beck. In many fitters, espedaltV ^e vOhsge controlled type, 
the amount of feedback is adjustable. The control for this 
purpose might be labelled: RESONANCE, EMPHASIS, or 
sometimes simply '^d" f'Q" H a factor used in measutln9 
ci^cuft resonances), but All of themptflfmthfrsfihe function. 

Fig. 5-1 1 shows a simplified block diagram of how feedback 
is applied inside a voltage controlled filter. For frequencies at 


Fig. 5-10 Positive Feedback in a PA System 



6 ooooO 


Amplifier 


Fig. S.11 VGF Retonenee Btetsk Df^lfeMt 



Resonance 

control 


Input 


> 



VCF 


Output 


Waveforms show the phase relatitmsforfmiuendesatthe 
cutoff point of the filter. 


62 Timbre 


Fls.S-12 VCF (Low Pass) Resonance 
(g) Small amount of mona nce 


(30 


dB -ID 


-JO 



10 lOM se 70 tsa «biaii M4dii tm u» moo mso 7i« nuoa 
Medium amount df mdiMTiea 


dB 


•10 

0 

























































































































1 

































-IB 
























































































































































































30 30 ■)»« an9»il»las im un son uoa tm> torn 


PREOUENCY 


(c) Large amount of resonance 

(But not enough to produce oscillations). 


dB 


*I0 
















































































































0 

-10 
-30 


















1 






























































































































































































































10 3030 airano ^ mt. ^ t/m tM a^no am ijm tofito 


64 Timbre 


S-10 Ringing In Filters. 

In Chapter One it was stated that: Resonance occurs when- 
ever a body or system is set into vibration at its own natural 
frequency as a result of impulses received from a body or 
system vibrating at the same frequency. Returning to the 
analogy of the child's swing, if the swing is hanging motionless 
and a single impulse or push is given to the swing, it will 
begin to swing at its natural frequency or period, after which 
it will slowly die to a motionless state again. A filter with 
resonaniM reacts in a similar way to impulses supplied by the 
waveform at the input of the filter. This type of reaction is 
called ringing. 

Fig. 5-14 shows what happens to a square wave apptfed to a 
filter with various degrees of resonance. Note that due to 
positive feedback, the vertical edges of the square wave 
overshoot their normal level, then waver up and down as 
they try to settle to the correct level. The frequency of 
tiiese wavarir^ is determined by the cutoff point of the 
filter. The wawform at (e) will of course occur whether 
there is an input waveform or not (the oscillations will 
begin as a rasult of internal circuit noise) and the frequency 
will be ^at of ^ cutoff firequency of the Alter. 

5-11 The Basic Synthesizer Patch 

Fig. 5-15 shows the completed basic synthesizer patch. 
This is the most common arrangement used to produce 

sounds. The variations possible on this patch are almost 
limitless. One of the most common variations is to use more 
*than one module or alemanl For mample, with nutnv 
sounds it is better to USB more than one VCO and/or mor« 

than one VCF. 

The VCF is a dynamic filter since, due to control voltages 
acfing on it. Its characteristics often change durfng the 
production of each note. Often it is desirable to simultane- 
ously ihflt^te the tone colors which do not change. This can 
be done by means of a itatlc fliter, a fitted whose character* 
istics remain fixed once it has been set by means of its front 
panel controls. Such a filter would normally be added between 
the VGF wid VGA. 


Fig. 5-14 Effects of Resonance on Waveform 
(a) Nomonanee 


® Ria^«M low 


(c) Resonance 
medium 


@ Resonance 
high 


Q Self 

oscillation 



fr = Cutoff frequency of filter 


Timbre 65 



66 Timbre 


5-12 QuMtilMii 

1. What K the diffsrancebMem a white noise ami #pmk^ ^ 

2. What are the four types of filter comnmnty u$ed in electronic music? What are their basic functions? 
3; How the tiutsff 0«HTft of s f^vst det^lmd^ 

4. What does it mean when we say a given filter has a fall off rate of -24dB/8va? 

5. Why is it necessary soUnattrntts to control the cutoff point of the filter with the keyboard control voltage? 

6. In a synthesize wi^ 1v/8va exponential VCCs, what voltage-to-frequency relation ^outd be used for the 
filfsf^Why? 

7. In a sound with a high amount of high pass filtering, the lower harmonics are effectively missing from the 
output sound. Why does the output sound stUI retain its sense of pitch? 

8. Make diagrams of the effects of low pass and high pass filtering on a square wave. 

9. How is resonance induced in a filter? How does it affect the filter's frequency response curve? How does it 
affect a square wave? 

10. What is ringing in a filter? What affect does filter ringing have on a square wave input? 


Words to define: 

band pass filter 
band reject filter 
BPF 

cutoff frequency 

feedback 

filter 

filter slope 
growl 

high pass filter 

HPF 

low pass filter 
LPF 

negative feedback 

noise 

oscillation 

pink noise 
positive feedback 
resonar«» (filter) 
ringing (filter) 
self-oscillation 
VCF 

voltage controlled filter 
white noise 


Metric Conversion Table 

The conversion table below allcM^ ^ifiek and eaiy comMoti from one metifG prefTx to another^ The prefixes 

listed are those which are commonly used in electronics and electronic music. To use the table, find the given 
prefix in the column on the left, then follow the line horizontally to the vertical column containing the desired 
pnt^ The figuif indicates the number of places the dedmid point must be movetf and the arrow thdibates the 

EXAMPLE: Convert 3 kHz to units (Hz). (3 kHz = 3 kilohertz.) Find "kilo" in the left-hand column; move to the 
right to the vertical column headed "units". 3 => indicates that the decimal point must be moved three places to 
the right, thus: 

3 kHz = 3,000 Hz 

EXAMPLE: Convert 380 milliseconds to seconds. Find "milli" in the left column: to the right, under "Units", 
readt^-S, tiius: 

3B0 mnilseeonck = 0.38 sisoond 


Metric Coiwariion Tiitole 


ORIGINAL 
VALUE 

DESIRED VAL 

JE 

Mega 

Kilo 

Units 

Deci 

Centi 

Milli 

Micro 

Micromicro 

Mega 


3=> 

6 => 

7 => 

8^ 

9=> 


18=> 

Kilo 

^ 3 



4=> 

5=^ 

6=> 

9=> 

15=* 

Units 

<- 6 

3 


1 =»- 

2=> 

3«* 


12* 

Deci 

*■ 7 

4-4 

^ 1 


1 ^ 

2* 

5=»- 

11«* 

Centi 

<= 8 

^ 5 

<= 2 

<= 1 


I-" 

4 => 

10=> 

Milli 

9 

6 

•= 3 

<= 2 

<= 1 


3 => 

9 

Micro 

<=12 

«= 9 

^ 6 

^ 5 

4 

^3 



Pico 


*«1S 

•0-12 


••=10 

^9 

6 



Index 


Additive lyn^Mtilft, flB^ 

ADSR,44 

AM. 47 

Amplifier, summing, 35 

voltage controlM, 20,44 
Amplify, 14 

Amplitude modulation. 47 
Anechoic chamber, 39 
Anilnode, 6 

Attacl< time (A), 44 
Attenuate, 14, 35 

Band pass filter (BPF), 52, 57 
Band reject filWR4.^,68 
Bar, 39 

Basic lynthesteAf mnitt ^ 38. 48, 65 
Beat, 29 

Beat frequency, 29 

Bell, Alexander Graham, 39 

BPF, 52, 57 

Cent, 32 

Central proceMlnS UAH tGPU},22 
Chord, 27 
Chorus effect, 10 
Clarinet, waveform of, 16 
Comprenion {cA str). 2 
Computer music, 20-24 
Concord, 27 
Consonance, 27, 30 
Control response of VGA, 45 

of VCF,55 

of VCO, 32-36 
Control voltage, 19 
Cutoff frequency, filter, 53 
Cutoff point, f lltw^, 63 

dB,39 

dB SPL,40 

Decay time (D), 44 

Decibel, 39 

Direct synthesis, 23 

Discord, 27 

Dissonance, 27, 30 

Dyck, Ralph, 22 

Dynamic filter, 84 

Dynamic mtmA%^ 

Dyhamt(^8,39 

Echo, 9 

Electronic music, 13-14 

live, 13 

recorded, 14 
Emphasis, 61 
End correction, 3 
Envelope, 8, 44 

organ, 44 

piano, 8, 44, 45 

synthesizer, 44 

vowel sound "ah", 8 
Envelope generator, 44 
Equal temperament, 28 
Even temperament, 28 
Exponential decay, 45 
Exponential generator, 34 
Exponential progression, 31-^2 
Exponential VCA, 46 
Exponentia) VCO, 33 


Feedback, negative, 60 

positive, 61 
Filter, 16, 51 

band pass, 52, 57 
- band reject, 52, 58 

cutoff frequency, 53 

frequency response of, 62, 56€9, 62 

fall off rate, 53 

dynamic, 64 

effects of resonance, 63 

effects on waveform, 58, 60 

high pass, 52, 56 

low pass. 52-56, 60 

pink noise, 53 

ringing in, 64 

slope, 53 

Btetlc.64 

voltage controlled, 19, S%.S6S$ 
Flute, spectrum of, 15 
waveform of, 1 6 

FM,36 
Frequency, 1 

Frequency modulation (FM), 36 
Frequency response of ear, 2, 40-41 

of filter, 52 
Fundfu^Bntal, 5 

Gate pulse, 44, 47 
Growl, 56 

Guitar, 6, 8 
Harmonics, 5, 29 
first, 5 

intensity of, 8,42 

inversion of, 18, 19 

second, 5 
Harmonic series, natural, 6, 30 
Harmony, 27 
Hearing, threshold of, 39 
Hertz (Hz), 2 
Hertz, Heinrich Rudolf, 2 
High pass filter, 52. 56-57 
Hold control (VCA), 47 
HPF,62,6667 

Initial gain, 47 
Intensity, 40 

of hamionics, 8, 42 
Interval, 27 

consonance and dissonance of, 30 

Just intonation, 28 

LFO,36,47, 56 
Linear scale, 31 
Linear VCA, 45 
Linear VCO, 32. 33 
Loudness. 8, 39, 40 

Low frequency oscillator (LFO), 36,47. 56 

control of VCA, 47 

control of VCF, 56 

control of VCO, 36 
Low pass filter (LPF), 52-56.60 

control response of, 55 

cutoff frequency, 53 

perfect, 53 

voltage controlled, 55-56 
voltage-to^requency relation, 55 
LPF {see Low pass flltier) 


Index 67 


Microbar (^ibar), 39 
MicroComposer, 22 
Microprocessor, 22 
Modulation, 19,36 

amplitude, 47 

frequency, 36 
Monochord, 3-4 
Music, computer, 20t24 

electronic. 13-14 

scale systems, 28 
Musique concrete, 1 3 

Natural harmonic series, 5, 30 
Negative feedback, 60 
Node, 6 

Noise, 20, 42, 51 
pink« 20,51. 53 

w ov f o i m of.go 

white, 20, 51 
Noise generator, 20, 42, 53 
Noti^Mmionie overtones. 7 

Octave, 27 
Oscillator, 16, 19,27 

exponential, 33 

linear, 32, 33 

low frequency, 36.47, 56 

voltage controlled, 19> 27 
Overtones, 4. 29 

non-harmonle, 7 

Patching, 19 
Pendulum, 14-15 
Phase, 18, 60 
Phase shifter, 10, 18 
Phon,41 

Pickup (gultBr),positton,;a^ 6 
Pink noise, 20.51,53 
Pink noise filter. 53 
Pipe, closed end, 3 

open end, 3 

scale of, 7 
Pitch, 1 , 27 
Polarity, 60 
Portamento, 60 
Positive feedback, 61 

f^toil«flBte,44^47 

0*61 

Ramp wave, 16 
Rarefaction, 2 
Releese time, (R ), 44 
Resonance, body Onsti'ument). 7 

dffined,2,64 

effect on waveform, 64 

frequency response In f ilttrs, 62, 63 

in closed end air eolur^Si'^ 

in filters, 61 

in open air cotumns, 3 

in strings, 3-4 
Resonance tubes, 3 
Response, filter frequency, 52 
Response, VCA control, 45 

VCF control, 55 

VCO control, 32-36 
RevertKration, 9 
Ringing in filters, 64 
Roland MC-8 MicroComposer, 22 


68 Index 


Sawtooth wave, 16-18, 
inversion of, 18 

spectrum of, 17, ■s' 
waveform of, 1 7 
Scale, comparison of linear 9(Kl'^N)0nentiat. 32 

Krwftr.SI 

musical, 28 
Scale of pipe, 7 
Self-oscillation, 61, 63 
Sine wave, 14, 60,63 

inversion of, 18 
Sonometer, 3-4 
Sound, 1 

defined, 1 

•peed of, 2 

three qualities of, 1 
Sound pressure level (SPL),40 
Sound waves, 2 

propagation of, 8 
Spatial effects, 10 
iG^ctrum diagrams, 16 

effecu of filtering, 64, 56. 63 

of clarinet, 16 

of flute. IS 

of sawtooth wave, 17, 

of square wave, 17, 43 

of triangle wave, 43 

of trumpet. 16 
SPL.40 

Bquare wave, 16, 42. 68 

inversion of, 18 

spectrum of, 17^^ 
Static filter, 64 

Subtractive synthesis, 16, 1&-20; S3t 
Summing amplifier. 36 
Sustain, (S),44 
Svntfiesis. 16, 19 

additive, 16 

direct, 23 

subtractive, 16. 19-20, 53 
$yfiimj»F, barie patch, ig, 35, 48, 65 
eofn^w ooncpol of, 22 
vdttaoB «onirolled, id 

Threshold of hearing, 39 
Threshold of pain, 39 
Timbre, 4, 51 
Tone color, 4, 51 

changes with pitch, 20, 55 
Transient generat!W,44 
Tremolo, 47 
Tittantf a wave, 42, 60 

spectrum of, 43 

wjaveform of, 43 
Trumpet, spectrum of, 15 

waveform of, 15 

TMningM^'t^a4,i4 


UrtiS(Hl,27 

VCA (see Voltage controlled amplifier) 
VCF isee Voltage controlled filter) 
VCO isee Voltage oottOe^ OldUator} 
Vibrato. 36 

V^t^^ntrolled amplifier (VCA), 2D, 44 
control by LFO, 47 

control response, 45 
exponential, 46 
linear, 45 

Voltage controlled filter (VCF). 19,51,55-56 

control by LFO, 56 

conttvl mponse, 56 

cutelf frnqiian^, 83 

frequency ravMHW,^ 
Voltage cbmir6llM«sdUfltt0)- fVCO), 19, 27 

control by LFO, 36 

control response, 32-34 

exponential, 33 

linear, 32, 33 
Voltage eontrollsci «|nnl»^>Mf^10 

ari^lituda niodiilMed. 47 
clarinet, 15 
effect of filtering, 58 
effect of resonance, 64 

electrical, 14 
flute, 1 5 

frequency nradulated, 36 
houshold electricity, 15 
f»l«,20 
Hiwtemh,i7 

sine wave, 14 

square wave, 1 7, 43, 60, 64 

triangle wave, 43 

trumpet, 15 
Wavelength, 2 
White noise, 20, 51